Systems and methods for dynamic audio processing

ABSTRACT

An audio processing system includes a server complex in communication with a network. The server complex receives a digital audio file and one or more analog domain control settings from a client device across the network. A digital-to-analog converter converts the digital audio file to an analog signal. One or more analog signal processors apply at least one analog modification to the analog signal in accordance with the one or more analog domain control settings. An analog-to-digital converter converts the modified analog signal to a modified digital audio file. The server complex can then deliver the modified digital audio file to the client device across the network.

CROSS REFERENCE TO PRIOR APPLICATIONS

This application is a continuation of, and therefore claims priorityunder 35 USC § 120 to, U.S. application Ser. No. 16/016,282, filed Jun.22, 2018, which is a continuation-in-part of, and therefore claimspriority under 35 USC § 120 to, U.S. application Ser. No. 15/188,795,filed Jun. 21, 2016, which is a continuation of, and therefore claimspriority under 35 USC § 120 to, U.S. application Ser. No. 14/224,009,filed Mar. 24, 2014, which is a continuation of, and therefore claimspriority under 35 USC § 120 to, U.S. application Ser. No. 13/209,368,filed Aug. 13, 2011, now U.S. Pat. No. 8,682,462, each of which isincorporated by reference for all purposes. U.S. application Ser. No.16/016,282 is also a continuation-in-part of, and therefore claimspriority under 35 USC § 120 to, U.S. application Ser. No. 15/257,642,filed Sep. 6, 2016, which is incorporated by reference for all purposes.

BACKGROUND Technical Field

The embodiments relate generally to systems and methods for processingaudio, and, more specifically, to systems and methods for processingaudio for increased perceived loudness while retaining changes inperceived volume.

Background Art

Audio production can include the pre-recording processing, recording,mixing, and/or mastering of sound. These phases of audio production canall involve processing of audio, which includes the manipulation of theaudio to produce an improved digital audio file.

During audio processing, a representation of audio can be manipulated(e.g., enhanced) as either a digital or analog signal. A digital signal(i.e., digital audio) comprises a series of ones and zeros thatrepresent a sound wave (i.e., audio). An analog signal (i.e., analogaudio) comprises a continuous electrical signal that represents thesound wave. Digital manipulation (i.e., modulation) involves processingthe ones and zeros of the digital signal, such as via a processorexecuting a formula. Analog manipulation (i.e., modulation) involvespassing the analog signal through one or more physical components, sucha circuit containing resistors, capacitors, op amps, and/or a vacuumtube. Whereas an analog compressor is made up of physical components, adigital compressor can be a set of instructions executed by a processor,such as a plug-in that operates within a digital audio workstation(DAW).

Typically, the audio that needs processing is one or more digital audiofiles. For example, a user may select one or more .WAV filesrepresenting songs that need processing. While the audio processing maytake place entirely in the digital domain, the digital audio is commonlyconverted to analog audio and manipulated with analog audio componentsin most commercial audio production environments. This is the case, inpart, because of the pleasing audio qualities that analog components canadd to the audio. However, in environments where cost is a factor, someor all of the audio production process may be carried out digitallythrough the use of plugins and software, some of which may attempt tomodel the characteristics of physical analog equipment.

For example, the recording process involves recording sound in thedigital domain in the form of digital audio files. Often, someprocessing, such as pre-mixing, of these files will occur in order toadd some clarity or change the levels of the recorded audio, and todetermine whether additional takes are necessary.

Similarly, the mixing process can involve processing audio by raising orlowering levels for particular tracks, adding effects, addingequalization, adding compression, and so forth, in order to create aclearer sounding audio production.

As another example, the mastering process involves enhancing recordedaudio from a source, such as a compact disc (CD) containing a final mixof the recorded audio, to create a master version of the audio withimproved sound translation and increased loudness for the best possibleplayback on various sound systems. The enhancement almost alwaysincludes modifying the audio by applying some form of compression,limiting, and/or equalization to the audio.

The end goal of the mastering process is typically to create a masterversion of the enhanced audio that can be used to replicate and/ordistribute the audio. For example, the master audio may be storeddigitally on a compact disk. Alternatively, an analog version of themaster audio may be stored on tape or vinyl. In either case, the mediumholding the final audio is referred to as the “master,” and is generallyused to replicate the audio, such as in the creation of vinyl, compactdiscs, digital files for download, or other music media for public use.

Mastering and mixing engineers and/or home users almost always need toapply corrective equalization and/or dynamics processing (e.g.,compression and/or limiting) in order to improve upon sound translationon all playback systems and increase loudness. When processing audio,dynamics processing (e.g., dynamic compression or limiting) is used toincrease the volume of the recorded audio to two or three times theoriginal volume so that the volume level can be competitive with that ofother music in the market for sale. Achieving competitive volume levelsis important so that the mastered song is not perceived as quieterand/or less energetic than other songs played on a listener's soundsystem. However, this type of dynamic enhancement usually flattens thevolume levels and dynamic changes in the audio, removing fluctuation indynamics (loud parts vs. quiet parts) so that the listener is less ableto distinguish volume changes in the music and the impact of dynamicinstruments like drums. This type of compression and limiting is verycommon and the increases in levels can also cause audible distortion inthe music.

Similar techniques are used, for example, to ensure that commercials areloud enough to stand out and catch the attention of viewers.Additionally, mixing engineers for television and movies process sounds,voices, music, etc. in order to achieve levels and clarity that isappropriate for the particular application.

In addition to audio professionals (e.g., mastering engineers, mixers,mixers for film (television and movie audio), audio engineers, audioproducers, recording studio engineers, studio musicians, etc.), homeenthusiasts and hobbyists may also be involved with various aspects ofaudio production. For example, some people record, mix, remix, master,and/or otherwise produce audio, such as music, as a hobby. Other peopleare stereo enthusiasts (e.g., audiophiles) who use hardware and/orsoftware to process “finished” audio to achieve a better listeningexperience. Production of audio at nearly any level involves some formof audio processing. However, these hobbyists and at-home enthusiastsare often limited by their lack of training and the expense required topurchase professional-level equipment for achieving commercial-levelloudness without destroying dynamics and/or introducing distortion.

Therefore, a need exists for systems and methods of processing audiothat can achieve commercially competitive audio levels withoutdestroying the dynamics (i.e., perceived volume changes) of the song orcausing distortion in the audio.

Accordingly, systems and methods are provided herein for processingaudio to bring the volume levels up to today's very loud digital levels(or louder) while reducing distortion and retaining more volume dynamics(i.e., perceived changes in volume) than prior systems have ever allowedin the past.

BRIEF DESCRIPTION OF THE DRAWINGS

The accompanying drawings, which are incorporated in and constitute apart of this disclosure, illustrate various embodiments and aspects ofthe present invention. In the drawings:

FIG. 1A is an exemplary illustration of a system for processing audio,in accordance with an embodiment;

FIG. 1B is an exemplary illustration of an alternate system forprocessing audio, in accordance with an embodiment;

FIG. 2 is an exemplary illustration of an audio processing device, inaccordance with an embodiment;

FIGS. 3A-B are exemplary flow charts with non-exhaustive listings ofsteps that may be performed in an audio processing environment, inaccordance with an embodiment;

FIG. 4 is an exemplary flow chart with a non-exhaustive listing of stepsthat may be performed by a digital audio workstation and an audioprocessing device that interface with one another, in accordance with anembodiment; and

FIG. 5 is an exemplary flow chart with a non-exhaustive listing of stepsthat may be performed by a digital audio workstation (DAW).

FIG. 6 illustrated one explanatory system in accordance with one or moreembodiments of the disclosure.

FIG. 7 illustrates one alternate server complex in accordance with oneor more embodiments of the disclosure.

FIG. 8 illustrates one explanatory method and system in accordance withone or more embodiments of the disclosure.

FIG. 9 illustrates one explanatory control device controlling one ormore analog signal processors in accordance with one or more embodimentsof the disclosure.

FIG. 10 illustrates additional explanatory control devices controllingone or more analog signal processors in accordance with one or moreembodiments of the disclosure.

FIG. 11 illustrates another explanatory system in accordance with one ormore embodiments of the disclosure.

FIG. 12 illustrates another explanatory system in accordance with one ormore embodiments of the disclosure.

FIG. 13 illustrates one explanatory method in accordance with one ormore embodiments of the disclosure.

DETAILED DESCRIPTION OF THE DRAWINGS

Before describing in detail embodiments that are in accordance with thepresent disclosure, it should be observed that the embodiments resideprimarily in combinations of method steps and apparatus componentsrelated to controlling, remotely, analog signal processors in anautomated analog domain mastering system to master a digital audio filein the analog domain. Any process descriptions or blocks in flow chartsshould be understood as representing modules, segments, or portions ofcode that include one or more executable instructions for implementingspecific logical functions or steps in the process. Alternateimplementations are included, and it will be clear that functions may beexecuted out of order from that shown or discussed, includingsubstantially concurrently or in reverse order, depending on thefunctionality involved. Accordingly, the apparatus components and methodsteps have been represented where appropriate by conventional symbols inthe drawings, showing only those specific details that are pertinent tounderstanding the embodiments of the present disclosure so as not toobscure the disclosure with details that will be readily apparent tothose of ordinary skill in the art having the benefit of the descriptionherein. Further, it is expected that one of ordinary skill,notwithstanding possibly significant effort and many design choicesmotivated by, for example, available time, current technology, andeconomic considerations, when guided by the concepts and principlesdisclosed herein will be readily capable of generating systems andmethods in accordance with the disclosure with minimal experimentation.

Embodiments of the disclosure do not recite the implementation of anycommonplace business method aimed at processing business information,nor do they apply a known business process to the particulartechnological environment of the Internet. Moreover, embodiments of thedisclosure do not create or alter contractual relations using genericcomputer functions and conventional network operations. Quite to thecontrary, embodiments of the disclosure employ methods that, whenapplied to analog signal processing circuits operating in tandem with aserver complex, allow digital control of analog domain equipment tomaster digital audio files in the analog domain remotely to obtain afinal mix.

Embodiments described herein include systems and methods for processingaudio. In one embodiment the, system comprises a processor that plays adigital audio file. The digital audio file may contain metadataspecifying a first clock frequency for normal playback. However, theprocessor plays the digital audio file at a second clock frequency thatis higher than the first (i.e., normal) clock frequency, resulting infaster than normal playback.

In one embodiment, a digital-to-analog converter converts the digitalaudio into an analog signal (representing analog audio) while thedigital audio is playing at the higher second clock frequency (i.e.,faster than normal). This may raise the low frequency information tobecome higher frequency information of the digital audio file duringplayback, as compared to playback at the first clock frequency. Thesystem may then pass the converted analog signal through an analogcircuit to manipulate at least one sound characteristic of the analogaudio. For example, the analog circuit may contain components forcompressing, limiting, and/or making equalization adjustments to theanalog audio.

Upon passing through the analog circuit, the system may route themanipulated analog signal to an analog-to-digital converter. Theanalog-to-digital converter may then convert the manipulated analogsignal into a manipulated digital audio file, which is stored on acomputer-readable storage medium. The processor then changes the clockfrequency associated with the modified digital audio file back to thefirst (i.e., original and normal) clock frequency, for normal playback.This can lower the frequency range of the modified digital audio file tofrequencies representative of the original digital audio file (asidefrom adjustments made using, for example, equalization duringprocessing).

In one embodiment, the audio processing is carried out across multipleworkstations and/or processors. For example, a first workstation mayoutput the digital audio file to an analog circuit, which in turnoutputs to a second workstation that converts the analog audio into amodified digital audio file. This may be thought of as a “throw andcatch” arrangement.

In another embodiment, the system includes a monitoring circuit thatconverts a segment of the modified analog audio into a preview segmentof digital audio that is played back for monitoring at the first clockfrequency prior to the creation of the entire modified digital audiofile.

In another embodiment, the manipulation of the digital audio file occursentirely within the digital domain.

It is to be understood that both the foregoing general description andthe following detailed description are exemplary and explanatory onlyand are not restrictive of the embodiments, as claimed.

Reference will now be made in detail to the present exemplaryembodiments, including examples illustrated in the accompanyingdrawings. Wherever possible, the same reference numbers will be usedthroughout the drawings to refer to the same or like parts.

During audio processing, a representation of audio can be manipulated,altered, or enhanced either in the digital domain or in the analogdomain. A digital signal suitable for processing in the digital domain,one example of which is a digital audio file, comprises a series of onesand zeros that can be converted into an analog sound wave defining auralaudio. By contrast, an analog signal suitable for processing in theanalog domain comprises a time-varying electrical signal that can drivea driver, such as a loud speaker, to create an analog sound wave.

Manipulation in the digital domain involves processing the ones andzeros of a digital signal. By contrast, manipulation in the analogdomain involves passing the analog signal through one or more analogsignal processors. Such processing components can include resistors,capacitors, inductors, operational amplifier, vacuum tubes, transistors,and other analog components. Whereas analog signal processors comprisephysical components, a digital signal processors can be reduced a set ofinstructions executed by a processor, such as a plug-in that operateswithin a digital audio workstation.

Modern recording studios frequently record audio in the digital domain.While some studios still use analog devices such as audio tapes, moststudios employ digital recording devices because storage, transmission,copying, and sharing a digital recording is far simpler than with anaudio tape. Illustrating by example, a studio may record a recording inthe digital domain and save it as a .WAV file. After the recording, the.WAV file will need to be mastered.

Mastering can take place entirely in the digital domain. For example, inenvironments where cost is a factor, some or all of the audio productionprocess may be carried out digitally through the use of plugins andsoftware, some of which may attempt to model the characteristics ofphysical analog equipment. However, as noted above, many artists,audiophiles, and purists prefer mastering in the analog domain due tothe superior acoustic characteristics that result in the final mix. Toaccomplish audio mastering from a digital file, the digital audio mustbe converted to analog audio and manipulated with analog audiocomponents. Analog mastering provides a more pleasing and rich sound dueto the way that audio signal processors raise or lower levels forparticular tracks, add effects, add equalization, add compression, andso forth to manipulate audio signals. The end goal of the masteringprocess is to create a master version of the enhanced audio that can beused to replicate and/or distribute the audio.

Mastering and mixing engineers and/or home users almost always need toapply corrective equalization and/or dynamics processing such ascompression and/or limiting to improve sound translation and loudness toensure proper fidelity on all playback systems. When processing audio,dynamics processing, e.g., dynamic compression or limiting, is used toincrease the volume of the recorded audio to two or three times theoriginal volume so that the volume level can be competitive with that ofother music in the market for sale. Achieving competitive volume levelsis important so that the mastered song is not perceived as quieterand/or less energetic than other songs played on a listener's soundsystem. However, this type of dynamic enhancement usually flattens thevolume levels and dynamic changes in the audio, removing fluctuation indynamics (loud parts vs. quiet parts) so that the listener is less ableto distinguish volume changes in the music and the impact of dynamicinstruments like drums. This type of compression and limiting is verycommon and the increases in levels can also cause audible distortion inthe music.

In addition to audio professionals such as mastering engineers, mixers,mixers for film, audio engineers, audio producers, recording studioengineers, studio musicians, home enthusiasts, and hobbyists frequentlyhave a need for mastering services. Illustrating by example, some peoplerecord, mix, remix, master, and/or otherwise produce audio, such asmusic, as a hobby. Other people are stereo enthusiasts or audiophileswho use hardware and/or software to digitally process “finished” audioto achieve a better listening experience. However, these hobbyists andat-home enthusiasts are often limited by their lack of training and theexpense required to purchase professional-level analog equipment forachieving commercial-level loudness without destroying dynamics and/orintroducing distortion.

Advantageously, embodiments of the disclosure provide a remote system bywhich users may not only master files in the analog domain, but alsocontrol the analog boards and analog signal processors to makeadjustments in accordance with their desired preferences. In short,embodiments of the disclosure allow users to remotely access theexpensive and complex analog signal processing devices of an analogmaster's highly controlled studio, and in particular, to control thoseanalog signal processing devices to master content from a computer,tablet, or phone without having to travel to the studio.

Embodiments of the disclosure allow remote, analog, audio mastering froma client device that is interfaced across a network with a servercomplex. A user employing a user interface at a client terminal canprovide one or more analog domain control settings from the clientterminal to control settings and other adjustments on one or more analogsignal processors located across the network to master files in theanalog domain. Using embodiments of the disclosure, a user can changethe settings on analog—as well as digital—equipment remotely. Parameterssuch as input level, output level, equalization, compression, and othersettings of analog or digital equipment can be adjusted using the clientdevice, which is located remotely from the server complex and analogsignal processing equipment.

Exemplary embodiments herein allow a user to create audio files that canbe perceived as louder, more dynamic, and/or less distorted than audiofiles created using traditional methods. In one embodiment, a digitalaudio file is assigned a clock frequency (i.e., second clock frequency)that is higher than the normal playback frequency (i.e., first clockfrequency). Then the digital audio file may be played at the higherclock frequency (resulting in a faster playing speed and higherfrequency information in the audio). In one embodiment, the digitalaudio file playing at the second frequency may then be converted to ananalog signal and processed using one or more analog equalizers and/oranalog dynamics processors (e.g., compressor, limiter, etc.) (i.e., ananalog circuit). The analog audio signal (playing at the faster speed)may then be converted to digital by an analog-to-digital converter. Inthe digital domain, further processing can be applied in one embodiment.The resulting modified digital audio file may be saved to anon-transitory computer-readable medium, where the clock frequency ofthe resulting digital audio file is reset to the original normalplayback frequency so that the modified digital audio file can be playedat its original speed.

By processing the digital audio file while it plays at the higherfrequency, less low frequencies are present in the digital audio file(and more high frequency information is present), and distortionattributable to passing low frequencies through digital-to-analogconverters, compressors, limiters, equalizers, and/or other componentsmay be reduced. This results in increased capability to make the audiolouder, which in turn can result in a louder, clearer, and more dynamicaudio file. Additionally, the faster playback speed can allow for fasteraudio processing when the entire analog audio signal must be convertedback into and stored as a modified digital audio file.

Consequently, an embodiment herein may help users (e.g., masteringengineers, television or film mixers (mixing for film), home stereoenthusiasts (audiophiles), and/or anyone else who processes audio)create audio files with a competitive volume without distortion ordiminished dynamics. For example, an embodiment also may help recordingstudios create listening versions of clients' recordings at acompetitive volume so that the recording may be much louder but notdistorted. As another example, a further embodiment may allow ‘at home’studio engineers to create competitive and quality sounding recordingswithout spending money on a mastering engineer. Because an embodimentmay allow for processing audio to contain higher volume levels withminimal difficulty, this may allow hobbyists and at-home enthusiasts ameans of creating commercially-acceptable productions with reduced costsand/or training.

The methods disclosed herein may be executed in full or in part, forexample, by a processor that executes instructions stored on anon-transitory computer-readable storage medium. Similarly, a systemdescribed herein may include a processor and a memory, the memory beinga non-transitory computer-readable storage medium. As used herein, anon-transitory computer-readable storage medium refers to any type ofphysical memory on which information or data readable by a processor maybe stored. Examples include random access memory (RAM), read-only memory(ROM), volatile memory, nonvolatile memory, hard drives, solid statedrives, CD ROMs, DVDs, flash drives, disks, and any other known physicalstorage medium.

Additionally, singular terms, such as “processor,” “memory,” and“computer-readable storage medium,” may additionally refer to multiplestructures, such a plurality of processors, memories, and/orcomputer-readable storage mediums. The same applies to the termcomputer, which is understood to contain at least one processor that iscommunicatively coupled to at least one memory.

As referred to herein, a “memory” may comprise any type ofcomputer-readable storage medium unless otherwise specified. Acomputer-readable storage medium may store instructions for execution bya processor, including instructions for causing the processor to performsteps or stages consistent with an embodiment herein. Additionally, oneor more computer-readable storage mediums may be utilized inimplementing a computer-implemented method. The term “computer-readablestorage medium” should be understood to exclude carrier waves andtransitory signals.

Additionally, although “mastering” may be used as an example throughout,it is understood that the following description applies to other formsof audio production and/or audio processing, such as mixing, recording,pre-recording, and other forms of post-production.

In one or more embodiments, a server complex is in communication with anetwork, such as the Internet. A remote device, which may be a computer,a tablet, a smartphone, or other device, receives a user interfacepresentation from the server complex. The user interface can include oneor more analog domain control settings, one or more digital domaincontrol settings, and a network upload portal. Using the user interface,the user can set the analog domain control settings and/or digitaldomain control settings to the desired level. Additionally, the user canupload a digital audio file through the network upload portal.

The server complex, which is in communication with the remote deviceacross the network, then receives the digital audio file, the one ormore analog domain control settings and/or the one or more digitaldomain control settings. Using analog signal processing as an example,the server complex then employs a digital-to-analog converter to convertthe digital audio file to an analog signal. A control device, which isoperable with one or more analog signal processors, then applies settingadjustments to the one or more analog signal processors in accordancewith the one or more analog domain control settings received from theremote device.

The one or more analog signal processors then apply at least one dynamicanalog modification to the analog signal. As the analog signalprocessors have been configured in accordance with the one or morecontrol signals, the at least one dynamic analog modification is appliedin accordance with the one or more analog domain control settingsreceived from the remote device. The application of the one or moredynamic analog modifications results in a conversion of the analogsignal to a modified analog signal. An analog-to-digital converter thenconverts the modified analog signal to a modified digital audio file.

The server complex can then share the modified digital audio file withthe user. For example, in one embodiment the server delivers a downloadportal, which facilitates download of the modified digital audio file tothe remote device across the network. Advantageously, the user neednever visit the studio or physically touch the analog signal processorsto control them as desired to master audio files in the analog domain.

Embodiments of the disclosure use various systems and methods forprocessing audio. In one embodiment, a system comprises a processor of aserver complex that plays a digital audio file. In one embodiment, adigital-to-analog converter converts the digital audio into an analogsignal (representing analog audio) while the digital audio is playing.

The digital audio file may contain metadata specifying a first clockfrequency for normal playback. In some embodiments, while playing thedigital audio file, the processor may optionally play the digital audiofile at a second clock frequency that is different from, i.e., higher orlower, than the first, i.e., normal, clock frequency. This results infaster than normal playback, and as will be described in more detailbelow, can be used to further augment the mastering process. However, itshould be noted that this adjustment of frequency is optional and inmany embodiments will be omitted. Where frequency adjustment isoptionally used, it may raise the low frequency information to becomehigher frequency information of the digital audio file during playback,as compared to playback at the first clock frequency.

Regardless of whether clock frequency adjustment is employed, in one ormore embodiments the system then passes the analog signal through one ormore analog signal processors to manipulate at least one soundcharacteristic of the analog audio. For example, the one or more analogsignal processors may contain components for compressing, limiting,and/or making equalization adjustments to the analog audio.

In one or more embodiments, a mixing console or other settingscontroller is operable to control the one or more analog signalprocessors. The mixing console includes potentiometers and othercontrols for combining, routing, and altering analog signals. The mixingconsole can adjust settings of the one or more analog signal processorsto change the volume level of the analog signal, the timber of theanalog signal, or the dynamics of the analog signal. The mixing consolecan also adjust the settings of the one or more analog signal processorsto combine or split the analog signals, such as from or to differenttracks, as well. The mixing console can control the one or more analogsignal processors to modify analog signals with one or more ofcompression, limiting, or equalization.

In one or more embodiments a control device, which can be any of arobotic arm, one or more digitally controlled relays, one or morevoltage controlled filters, one or more digitally controlledservo-driven potentiometers, one or more digitally controlledservo-driven attenuators, one or more digitally controlled voltagecontrolled amplifiers, one or more digitally controlled variable gainamplifiers, or combinations thereof, are operable to apply receivedanalog domain control settings to the mixing console. For example, if auser uploading a song into a web portal wants increased compression,they can indicate this by transmitting analog domain control settings tothe server complex. Where the control device is a robotic arm, therobotic arm can adjust the knobs, sliders, and/or potentiometers toapply the increased compression settings to the analog signalprocessors. Advantageously, this allows the user to control the mixingconsole and its corresponding analog equipment remotely.

Upon passing through the analog circuit, the system may route themanipulated analog signal to an analog-to-digital converter. Theanalog-to-digital converter may then convert the manipulated analogsignal into a manipulated digital audio file, which is stored on acomputer-readable storage medium. Where the optional clock frequencymodification described above was included, the processor may then changethe clock frequency associated with the modified digital audio file backto the first, i.e., original and normal, clock frequency, for normalplayback. Where employed, this can lower the frequency range of themodified digital audio file to frequencies representative of theoriginal digital audio file. However, as noted above, clock frequencyadjustment is optional and will not be included in one or moreembodiments.

In one or more embodiments, the audio processing is carried out acrossmultiple workstations and/or processors. For example, a server complexmay include a first workstation to output the digital audio file to theone or more analog signal processors, which in turn outputs an analogsignal modified in accordance with received analog domain controlparameters to a second workstation. The second workstation may thenconvert the modified analog signal into a modified digital audio file.This may be thought of as a “pitch and catch” arrangement. In otherembodiments, the server complex may be simpler in structure. Forexample, rather than having two workstations, a single workstation mayinclude multiple outputs and/or inputs to convert a digital file to ananalog signal, deliver the analog signal to the one or more analogsignal processors, and then receive the modified analog signal back fromthe analog signal processors. Other configurations of the server complexwill be obvious to those of ordinary skill in the art having the benefitof this disclosure. It is to be understood that both the foregoinggeneral description and the following detailed description are exemplaryand explanatory only and are not restrictive of the embodiments, asclaimed.

FIG. 1A is an exemplary illustration of a system 100 for processingaudio, in accordance with an embodiment. In this example, the componentsof the system are split into the digital domain 110 and analog domain160.

In particular, the system 100 may include a computer (e.g., workstation)115 that stores a digital audio file 138. The computer may comprise oneor more computers (e.g., workstations). A workstation (e.g., digitalaudio workstation (DAW)) can comprise at least one processor and acomputer readable storage medium. In one embodiment, the workstation isa stand-alone device built specifically for handling audio production,mixing, and/or processing. For example, the workstation may have anintegrated mixer, audio sequencer, and/or effects capabilities. Inanother embodiment, the workstation can comprise a personal computerwith software being executed by a processor for the purpose of audioproduction, recording, mixing, and/or mastering.

In one embodiment, the digital audio file 138 is stored when thecomputer 115 records the digital audio file (e.g., in a recordingenvironment). In another embodiment, the computer 115 may simply importand store a previously-recorded digital audio file 138. For example, ata mastering studio, a client may bring a CD containing the digital audiofile 138, which is then accessed by computer 115. Alternatively, theclient may provide a link for downloading the digital audio file 138onto computer 115, such as by sharing a cloud-computing foldercontaining the digital audio file 138.

The digital audio file 138, as discussed herein, may include any fileformat that contains a representation of audio, such as .WAV, .AIFF,.MP3, SDII, AC3, DSD, or any number of audio file formats. For example,the digital audio file 138 shown in FIG. 1A is a .WAV file, which iscompatible with the Windows™ operating system and typically containsnon-compressed audio information (i.e., a relatively large file thatcontains all recorded audio information). However, other file types arepossible. For example, the digital audio file 138 can even include avideo file type, such as .AVI, to the extent that the video file typeincludes an audio track or portion.

The digital audio file 138 may also contain metadata that specifiescharacteristics of the digital audio file 138, such as the bit rate andthe sample rate. Other characteristics can also be identified in themetadata. For example, .WAV files contain a header that can indicatesurround sound and speaker positions, provide information regardingsample types, and supports defining custom extensions to the formatchunk.

The sample rate may indicate the number of samples per second used in adigital representation of an analog signal. The bit rate may indicatethe number of bits used to represent the level of the sample. In theory,the higher the sample rate and bit rate, the closer a discrete digitalaudio file represents the continuous analog audio signal that itemulates.

The normal playback or recording frequency (sample rate) can varybetween different digital audio files. The playback frequency is thesample rate indicated by the metadata, in an embodiment. For example,the standard sample rate (i.e., normal playback frequency) used fordigital audio files on audio compact discs (e.g., music CDs) is 44,100samples per second (44.1 kHz), with 16 bits of information per sample.Digital Video Discs (DVDs), on the other hand, contain digital audiofiles with a sample rate of 48 kHz and 24 bits of information persample.

For example, to playback a digital audio file recorded at 44.1 kHz, theplayback device will either read the metadata and automatically switchto a 44.1 kHz sample rate, or the user may have to select what samplerate the audio was recorded at, depending on the embodiment. If thewrong sample rate is selected the audio may playback at an incorrectspeed. Some systems may automatically sample rate convert the digitalaudio if the correct sample rate is not selected in the system. Thiswill resample the audio file so that it plays at the correct speed(maintaining the frequencies of the originally-recorded audio). Samplerate conversions generally can lead to fidelity loss and are avoided byaudio professionals if possible.

Additionally, music files can be recorded at a variety of differentsample rates (resulting in a variety of different normal playbackfrequencies). For example, some professional audio hardware provides theoption for sample rates of 88.2 kHz, 96 kHz, and/or 192 kHz. Even thoughstandard audio applications tend to call for digital audio files with44.1 kHz or 48 kHz sample rates, higher sample rates can be useful inaudio recording applications where effects are applied to ensure thatthe modified source information is as close to the original analogsignal (e.g., the signal generated by pressure on a microphonediaphragm) as possible.

In the case of audio with a non-commercial sample rate, the sample ratecan be converted to a standard sample rate (e.g., 44.1 kHz or 48 kHz) ata later time, such as when creating mixes or master versions of theaudio. Converting the sample rate involves re-approximating therepresented audio signal at the new sample rate, in order to preservethe frequencies and overall sound of the digital audio file. This is adifferent concept than changing the playback frequency of a digitalaudio file, which causes the digital audio file to play back faster orslower at higher or lower frequencies, respectively. Converting thesample rate instead maintains the frequency response of the audio.

In one embodiment, the digital audio file 138 may have metadataindicating a first clock frequency to use for normal playback. Forexample, the metadata may indicate a sample rate of 44.1 kHz. In oneembodiment, the sample rate may also be the clock frequency.

In another embodiment, the sample rate can be extrapolated into a clockfrequency to use for normal playback. For example, because each sampleof the digital audio file contains multiple bits-worth of information,if the system ties the clock to a particular amount of data to beprocessed, the actual clock frequency for playback may also depend onthe bit rate, which also may be indicated by metadata in the digitalaudio file 138. However, the sample rate indicated by metadata in manysystems indicates the actual clock frequency for normal playback,eliminating the need for the processor to calculate a different clockfrequency for use in playback. However, either embodiment is consideredto indicate a first clock frequency for a processor to use for normalplayback.

In one embodiment, the digital audio file is converted to an analogsignal. The processor (e.g., of the digital audio workstation 115)facilitates playback by routing information from the digital audio file138 at a specified playback clock frequency (e.g., sample rate). In oneembodiment, the information is routed to a digital-to-analog converterby the processor. The digital-to-analog converter converts the digitalsignal into an analog signal (used in analog domain 160), which isultimately supplied to speakers to produce pressure differences in theair that are perceived as sound.

In another embodiment, the processor routes the information to a digitalprocessor module (e.g., plugin) that emulates analog hardware. This canallow for additional digital effects to be applied to the digital audiofile 138 in the digital domain 110 in a way consistent with how effectsare applied in real time in an analog domain 160. However, the digitalaudio is not audible to a listener without first being converted into ananalog signal.

In one embodiment, the processor is included in computer 115 (which caninclude one or more computers). In another embodiment, the processor islocated outside computer 115, such as in an interface or module that iscommunicatively coupled to computer 115.

The processor may cause the audio file 138 to play at a second clockfrequency that is higher than the first clock frequency. For example,the processor may set the metadata of the digital audio file to indicatea second clock frequency for playback that is double the first clockfrequency. However, other combinations are possible, such as a 25percent higher clock frequency.

Using the second (i.e., higher) clock frequency for playback causes thedigital audio file 138 to playback at a faster speed than normal. As aresult, the digital file exhibits higher frequency characteristics thanwhen played at the normal playback frequency, and also completesplayback sooner. For example, by doubling the clock frequency, a digitalaudio file with audio information up to 22,500 Hz can have audioinformation up to 44,100 KHz, which is far outside the range of humanhearing.

In one embodiment, the clock frequency is chosen to substantially reduceor virtually eliminate audio frequency information below 250 Hz. Thislow frequency information often creates a “muddy” sound and may be thecause of distortion created by digital-to-analog converters and/oranalog components, such as a compressor, or digital components, such asdigital audio processors. The exact clock frequency needed to raise thelow frequency information above this threshold may vary depending on thesource audio information. For example, if an audio file has substantialaudio information at 200 Hz, a 25 percent increase in clock frequencywill move that audio information to above 250 Hz. In one embodiment, theideal clock frequency is chosen automatically by the processor, whichanalyzes the digital audio file to determine which clock frequency willmove audible levels of audio information to above 100 Hz.

In one embodiment, a digital-to-analog converter may convert the digitalaudio into an analog audio signal while the digital audio is playing atthe higher second clock frequency. Because the processor plays thedigital audio file 138 at the higher second clock frequency, less lowfrequency information is passed through the converters (relative to whenthe digital audio file 138 is played at a higher frequency), which mayreduce distortion and allow for a louder analog audio signal.Eliminating and/or reducing low frequency information lightens the loadon these components (including the analog input of analog-to-digitalconverters, in which low-frequency information can account forsignificant portions of current, causing overloading and/or distortion),which can result in a clearer analog audio signal.

The audio signal, as a result, may require less compression since it isalready louder. This, in turn, may also allow for maintaining dynamicsin volume while still achieving commercial loudness levels. This mayfurther lead to more clarity in the digital-to-analog conversion, sincelower frequencies are often the cause of the most audible distortionduring the conversion process.

The digital-to-analog converters may reside on computer 115 in oneembodiment, for example, as part of a sound card. In another embodiment,the converter(s) may be located externally to computer 115.

The clock signal used to play the digital audio 138 at the higher secondclock frequency may be generated by computer 115 (e.g., by theprocessor) in one embodiment. Alternatively, a module communicativelycoupled to the computer 115 may be responsible for generating the clocksignal in another embodiment. For example, a separate clock module maybe used to reduce an effect called jitter by having the clock modulesupply the processor with a more accurate clock signal. Other modules,such as the digital-to-analog converter module, may alternatively supplythe clock signal to the processor.

Although the sample rate is changed in metadata to reflect the secondclock frequency in one embodiment, an alternate embodiment does notalter the metadata. Instead, the DAW 115 may notify an externalconverter of the playback clock frequency to use. Or the user may selectthe clock frequency on the device supplying the clock signal. Theexternal converter may not check the metadata of the digital audio file,but instead will supply the clock at the frequency indicated by the DAWor user. In this embodiment, after the audio has been processed, theresulting modified digital audio file may already contain the correctmetadata for sample rate. However, in one embodiment, the externalconverter must be notified to change the clock frequency back to thefirst frequency for normal playback.

In one embodiment, once the digital signal (created in the digitaldomain 110 during playback) is converted to an analog signal, an analogcircuit 162 may apply at least one dynamic modification to the analogsignal. The dynamic modifications (i.e., effects) applied may include atleast one of compression, limiting, and equalization. In one embodiment,additional effects are possible, such as stereo field effects, excitereffects, tape emulation effects, etc. In order to apply these effects,the analog circuit 162 may comprise one or more hardware modules, suchas modules 165, 170, and 180. The modules may comprise any knowncombination of circuitry and analog components for applying compression,limiting, and/or equalization, depending on the dynamic effect appliedby the particular module.

Additionally, each of the modules may be connected to one another in aneffects chain in one embodiment. In an effects chain, the output fromone module can serve as an input for another module. For example, acompressor module 165 may output a modified analog signal that isreceived as an input at a limiter module 170. The output of limitermodule 170 may then be received as an input of equalization module 180.In the example shown in FIG. 1A, the output of the equalization module180 could be sent to an analog-to-digital converter so that the modifiedanalog signal may be converted back into digital audio. Additionally,although the example in FIG. 1A illustrates a signal chain whereincompression is provided first, then limiting, and then equalization,effects may also be provided in other orders. For example, equalizationmay be applied before any compression in another embodiment.

In one embodiment, multiple modules of the analog circuit 162 may bepart of a single hardware module (e.g., product) that is capable ofapplying multiple effect types.

Continuing with the example of FIG. 1A, the compressor module 165 isused to compress the dynamic range of the audio signal. This type ofcompression is distinct from data compression, in which the informationis optimized for a smaller file size. In dynamic range compression,quiet sounds can be made louder by reducing the dynamic range ofloudness and amplifying the quiet sounds.

The type of compression applied may vary between embodiments. Forexample, a peak sensing compressor may respond to an instantaneous levelof the input signal. This type of compression may provide tighter peakcontrol, but can yield very quick changes in gain reduction, which undertraditional audio processing methods can lead to audible distortion.Alternatively, an averaging compressor may be used to apply an averagingfunction (such as root mean squared (“RMS”)) on the input signal beforeits level is compared to the threshold. Some compressors may includecontrols or inputs to set a compression ratio, which typicallydetermines the reduction of signal loudness, and a gain level toincrease the loudness of the audio signal. Other controls, such asattack, release, and knee control may be provided to help shape thecompression. The attack may determine the period when the compressordecreases gain to reach the level governed by the ratio. The release maydetermine the period when the compressor is increasing gain to the levelgoverned by the ratio, or, to zero dB, once the level has fallen belowthe threshold. The length of each period may be determined by the rateof change and the required change in gain. In one embodiment, the attackand release times are adjustable by the user. In another embodiment, theattack and release times determined by the circuit design and cannot beadjusted by the user.

In an embodiment, providing an audio signal at a second (i.e., higher)clock frequency reduces the distortion caused by the compressor module165. This is because less low frequencies may be presented to thecompressor module 165 than if the signal had been created by playing theaudio at the first clock frequency. Because lower frequencies can causea bottle neck in compressors, restricting how much output can beattained before distortion occurs, providing a signal with less lowfrequency information can result in less distortion when applyingcompression. Continuing with FIG. 1A, a limiter module 170 may receive amodified analog signal from compressor module 165. Limiting, as providedby the limiter module 170, is technically another form of compressionthat includes a very high compression ratio. For example, a compressionratio between 60:1 and .infin.:1 may be used in limiting. The purpose oflimiting is generally to keep the audio signal level below 0 dB, toavoid “clipping.” Audio engineers and producers typically try to avoidclipping because clipping results in a harsh and typically undesirableaudio artifact. In an alternate embodiment, limiting is not appliedbecause the converters effectively limit the audio signal when thelow-frequency information is no longer present.

With prior systems, if limiting is relied on too heavily to reduce audiolevels, overload and distortion can occur. For example, when the signalprocessed by the limiter is consistently far above 0 dB, the amount ofcompression applied by the limiter can cause distortion for similarreasons as explained above with regard to compressors. But, in oneaspect, because the analog signal is created at the higher second clockfrequency, less low frequencies may be presented and outputted to andfrom the limiter module 170 than if the signal had been created byplaying the audio at the first clock frequency. Providing a signal withless low frequencies, as accomplished in an embodiment herein, canresult in less distortion during limiting.

As shown in FIG. 1A, an equalization module 180 may apply equalizationto the audio signal. Equalization may alter the frequency response ofthe audio signal, amplifying some frequencies and/or reducing somefrequencies. This can be used, for example, to emphasize differentfrequencies across the stereo field to make particular sounds,instruments, and/or voices stand out in an audio mix. However, analogequalization hardware, particularly cheap equalization hardware commonlyfound in home studios, can introduce distortion in the low frequenciesif the audio signal is too loud for the equalizer to handle. Therefore,by using an audio signal generated according to a second (i.e., higherclock frequency), less low-end frequency information is effected by anysuch distortion.

In the example of FIG. 1A, once the modified analog signal is outputfrom the last effects module (e.g., equalization module 180), themodified analog signal is converted back into a digital audio filethrough use of an analog-to-digital converter. This conversion occurswithout changing the speed of the audio file. In other words, theconverted file initially may be set to play at the second clockfrequency.

In an alternate embodiment, the analog-to-digital converter may be setto change the playback clock frequency (e.g., sample rate) of themodified audio signal as compared to the original digital audio filewithout modifying the metadata. In this instance, the playback clockfrequency supplied (e.g., using a crystal oscillator) by theanalog-to-digital converter may be changed accordingly to cause themodified digital audio file to play at the same speed as the originaldigital audio file with the second (i.e., higher) playback frequency. Inone such embodiment, the external converter may not know the contents ofthe metadata at any point in the process. In this way, no changes to thesample rate specified in metadata occur in one embodiment.

The resulting manipulated digital audio file is stored on anon-transitory computer-readable storage medium in one embodiment. Thisnon-transitory computer-readable storage medium may be located oncomputer 115 in one embodiment, such as on a disk drive or some otherstorage medium. In another embodiment, the non-transitorycomputer-readable storage medium is located on a separate product orworkstation from computer 115.

Once the manipulated digital audio file has been stored, in one aspect,the processor sets the metadata of the manipulated digital audio toindicate the first clock frequency for normal playback speed. Thiseffectively restores the frequency response of the manipulated digitalaudio file heard when the manipulated digital audio file is played,eliminating any “chipmunk effect” caused by setting the playbackfrequency to the second (i.e., higher) frequency prior to dynamicenhancement.

The processor that sets the metadata of the manipulated digital audiocan be one or more of the processors included in computer 115 in oneembodiment. However, because the term “processor” can include aplurality of processors, including processors that are part of differentdevices and/or workstations, the processor that sets the metadata of themanipulated digital audio to indicate the first clock frequency fornormal playback speed may be located somewhere besides computer 115,such as in a different workstation or device in one embodiment.

In an alternate embodiment, one or more of the analog modules 165, 170,and/or 180 are modeled in the digital domain 110. “Modeling” may includea series of algorithms or equations that emulate the effect of hardwareused in the analog domain 160 to manipulate the analog signal. Forexample, each component of a compressor module 165 may be modeled suchthat a digital effect can be created that functions similarly to theanalog counterpart. Rather than applying a particular dynamicenhancement in the analog domain 160, the modeled digital effect isinstead applied in the digital domain 110. In this alternate embodiment,digital effects modules may be employed to emulate one or more analogmodules 165, 170, 180. For example, the digital audio file 138 may stillbe played at the second (i.e., higher) frequency, during which time thedigital effects are applied to the digital audio signal. Because thedigital signal is supplied to the emulated analog circuit at the second(i.e., higher) clock frequency, results similar to those described withrespect to the analog domain 160 may be possible.

FIG. 1B is an exemplary illustration of an alternate system 190 forprocessing audio, in accordance with an embodiment. This alternateembodiment utilizes multiple workstations 115 and 185 to carry out theaudio processing. Each workstation 115 and 185 can include its ownprocessor(s). It is understood that reference to a processor herein caninclude both a first processor of the first workstation 115 and a secondprocessor of the second workstation 185.

In the illustrated system 190, a first workstation 115 may convert theoriginal digital audio file 138 a into analog, and send the analogsignal to the analog circuit 162 for processing. It is understood thatthis conversion can utilize an external converter in one embodiment.

Then, the modified (i.e., processed) audio is sent to the secondworkstation 185. In one embodiment, this includes sending the modifiedanalog signal to the second workstation 185, where it is converted intoa modified digital audio file 138 b. In this embodiment, the modifieddigital audio file 138 b can be stored on the second workstation 185 oron some other computer-readable medium.

The other aspects the system 190 in FIG. 1B can behave similarly toembodiments described with respect to FIG. 1A.

Turning to FIG. 2, an exemplary illustration of an audio processingdevice 205 is shown in accordance with an embodiment. Audio processingdevice 205 may be a standalone product in one embodiment, that connectsto a DAW.

The audio processing device 205 may receive audio information throughinput 210 a in one embodiment. This audio information may be a digitalaudio file. The digital audio file may be a portion of some largerdigital audio file in one embodiment. For example, the audio processingdevice 205 may interface with an external digital audio workstation(DAW) in one embodiment, and receive a portion of a digital audio filefrom the DAW. In one embodiment, the DAW may be executing audiosequencing and/or editing software that allows a user to select aportion of a digital audio file for manipulation. Software executed onthe DAW, such as a plugin, may facilitate communications between the DAWand the audio processing device 205 such that the workstation may exportat least a portion of a digital audio file to the audio processingdevice 205. These communications may be received by the audio processingdevice 205 through digital input 210 a. The protocol for communicationscan vary between embodiments. The DAW is able to automate the audioprocessing from within the DAW software environment in one embodiment,sending one or more commands to the audio processing device 205 tocontrol various aspects of the mastering process.

In one embodiment, the audio processing device 205 also contains anoutput 210 b for sending the manipulated digital audio file to areceiver device. The receiver device can be the DAW, or it can be someother device that includes a non-transient computer-readable storagemedium. In one embodiment, the DAW causes the audio processing device205 to export a manipulated digital audio file back to the DAW. Once theDAW receives the manipulated digital audio file, the DAW mayautomatically integrate the manipulated digital audio file into the DAWenvironment. For example, if the DAW is used in a movie productionenvironment, an audio portion of a video file may be sent to the audioprocessing device 205, and the DAW may automatically replace the audioportion with the manipulated digital audio file received from the audioprocessing device.

In another embodiment, the manipulated digital audio file is stored on acomputer-readable storage medium contained in the audio processingdevice 205, and is manually exported later, such as by connecting theaudio processing device 205 to a DAW and browsing memory contents forthe manipulated digital audio file. In this embodiment, the contents ofthe computer-readable storage medium contained in the audio processingdevice 205 may be browsed. In one embodiment, the contents are browsedfrom the DAW. In another embodiment, display 215 is capable ofdisplaying files 250 currently stored on the audio processing device205.

In one embodiment, a single cable connects the audio processing device205 to the DAW, through a single connection that encompasses both input210 a and output 210 b. This connection may be a transceiver. Theembodiments discussed herein are not limited to a specific transferprotocol. For example, USB, Firewire, Ethernet, HDMI, SATA, and SAS arejust some of the protocols that may be implemented in variousembodiments to facilitate communication and file transfers between theaudio processing device 205 and a DAW.

In one embodiment, a first level control 212 a is provided to controlthe volume level of the audio. In one embodiment the level control 212 acontrols the level of the analog audio signal before it is routedthrough the analog circuit. In another embodiment, the level control 212a controls the level of the received digital audio file. This may allowa user to raise or lower the volume of the analog or digital audio filebefore it is manipulated by audio processing. Similarly, in anotherembodiment, an output level control 212 b is provided for adjusting thevolume of the manipulated analog audio before it is sent to ananalog-to-digital converter. In another embodiment, the output levelcontrol 212 b controls the level of a manipulated digital audio file(i.e., after audio processing), before the manipulated audio file issent back to the DAW.

In one embodiment, the audio processing device 205 may containdigital-to-analog converters for converting the digital audio file intoan analog signal before audio processing. Similarly, the audioprocessing device 205 may contain analog-to-digital converters forconverting the manipulated audio back into a manipulated digital audiofile. In addition or alternatively, the audio processing device 205 maycontrol the DAW to cause the digital-to-analog conversion andanalog-to-digital conversion to occur using the converters used by theDAW. For example, if the DAW is already equipped with and/orcommunicatively coupled to high-end converters, it may be advantageousto use those DAW converters instead of converters that may be built intothe audio processing device 205. In still another embodiment, the audioprocessing device 205 is equipped with an interface for connecting toexternal converter modules. This may allow the audio processing device205 to utilize stand-alone converters for the conversion process. Theinterface can use any protocol known in the art for communicating withD/A and A/D converters.

In the example of FIG. 2, the audio processing device 205 contains adisplay 215 for assisting the user in applying various dynamicadjustments to the audio. The display 215 can be a liquid crystaldisplay in one embodiment. In another embodiment, the display 215 can bea touch screen display. In one embodiment, the display helps the usercontrol the analog circuit for applying compression 240, equalization242, and/or limiting 244.

Additionally, in one embodiment, the audio processing device 205 alsoallows the user to specify the second clock frequency that is used forplaying the digital audio file at a faster speed during conversion intoan analog audio signal. In one embodiment, the second clock frequencymay be selected based on a multiple of the original (i.e., normal)playback clock frequency. For example, as shown in FIG. 2, the user mayselect to double the playback clock frequency, which results in doublingthe frequency response characteristics of the digital audio file, andcauses playback to occur at two times the normal playback speed. In thisembodiment, the audio processing device 205 may automatically detect thefirst (i.e., normal) clock frequency of the digital audio file. This canbe done, for example, by recognizing the file type, determining theclock frequency metadata that corresponds to that file type, and thenretrieving the first clock frequency from the metadata. For example, ifthe first clock frequency is 44.1 kHz, the second clock frequency in theexample of FIG. 2 could be 88.2 kHz. In another embodiment, the user mayenter a specific clock frequency to use as the second clock frequency.

The audio processing device 205 may automatically store the first (i.e.,normal) clock frequency of the digital audio file, so that themanipulated digital audio file can be restored to the first clockfrequency after the manipulated digital audio file is created. In thisembodiment, once the manipulated digital audio file is created (e.g.,after manipulation by the analog circuit and conversion by theanalog-to-digital converter), the audio processing device 205 may setthe clock frequency value in the metadata of the manipulated digitalaudio file to indicate the first clock frequency. Thus, when themanipulated digital audio file is played back on the DAW (or, in oneembodiment, on the audio processing device), the playback will soundnormal and not have the added “chipmunk” effect.

Additionally, in one embodiment, the audio processing device 205 maysend 246 the analog audio signal to external analog devices. Forexample, output 248 a may be used to couple the audio processing device205 with an external analog device. The analog audio signal can then besent, for example, to a compressor, limiter, and/or equalization modulethat resides external to the audio processing device 205. A return 248 bmay be provided for returning the manipulated analog signal back to theaudio processing device 205.

In still another embodiment, the audio processing device 205 may containa monitoring output for listening to the audio during the masteringprocess. In general, this allows a user to hear the effects of themastering and make adjustments to the various modules (i.e., components)of the analog circuit. In one embodiment, the user may listen to thesped-up audio during manipulated by the analog circuit.

In another embodiment, a “time warp” monitoring feature is used formonitoring the analog audio at the normal playback frequency. This mayallow a user to listen to the audio without the “chipmunk” effect, andhear how the audio will sound once the playback clock frequency is resetto the first frequency. In one embodiment, the audio processing device205 may utilize a second pair of analog-to-digital converters to createshort digital audio files (i.e., monitoring files) that representsegments of the analog audio signal being manipulated. The processor maythen set the playback frequency of the short digital audio files to thefirst clock frequency, effectively slowing the playback speed to normal.The short digital audio files may then be played in succession byconverting them back into an analog signal that is sent to monitors(e.g., speakers and/or headphones).

These short audio files may range in length in various embodiments. Inone embodiment, the short digital audio files are 5 seconds long. Theaudio processing device 205 may create these short monitoring files, forexample, by converting an even shorter segment of analog audio to adigital audio file, setting the playback frequency to the slower firstfrequency. Although this technique necessarily will cause monitoring tolag a few seconds behind any dynamics modifications applied by the user,it may still allow the user to listen to segments of audio without thechipmunk effect, so that the user does not need to complete themastering process before hearing the results of the dynamicsmodifications at the normal playback frequency.

Additionally, in one embodiment with the “time warp” monitoring feature,the user can select the length of the monitoring segments. While longerlengths may allow a more natural listening experience (i.e., lesschopped up audio segments), more time will lapse between when the usermakes a dynamic adjustment (i.e., compression, limiting, and/orequalization) and when the user can actually hear the result of theadjustment at the first playback frequency.

The analog audio segments that are converted for monitoring purposes maynot be continuous. This is because audio files play slower at the firstclock frequency than at the second clock frequency. Therefore, to ensurethat the monitoring does not lag too far behind the dynamics adjustmentsmade by the user, the time interval (e.g., 5 second) specified by theuser may be used to “catch up” the monitored files, such that at thebeginning of each time interval a new monitoring segment begins nearreal time. With this method, monitoring segments of shorter lengths,such as 1 second, may allow for monitoring near real time, but withchoppy playback since each segment begins near real time based on themanipulated audio signal, which is playing at a faster speed based onbeing created with the second clock frequency.

In one embodiment, the user may manually select whether the monitoringoutput is real time monitoring of the sped up playback, or time warpedmonitoring of the manipulated audio signal.

FIGS. 3A-B are exemplary flow charts with non-exhaustive listings ofsteps that may be performed in accordance with an embodiment. At step812 a of FIG. 3A, a DAW (e.g., computer, audio processing device, etc.)receives a digital audio file having a first clock frequency for normalplayback. The digital audio file may be received by importing the filein one embodiment. The digital audio file may also be received byrecording audio onto a computer-readable medium.

As step 812 a of FIG. 3A indicates, the digital audio file has a firstclock frequency associated with it for normal playback. Similarly, instep 812 b of FIG. 3B, a processor plays a digital audio file, whereinthe digital audio file has metadata indicating a first clock frequencyto use for normal playback. Because digital audio files can be createdusing various different sample rates, the first frequency for normalplayback may correlate to the sample and bit rate of the particulardigital audio file. Normal playback includes playing the audio file backwithout changing the pitch or the speed of the digital audio file.

At step 812 b, the processor causes the digital audio file to play at asecond clock frequency that is higher than the first clock frequency.Similarly, at step 814 a, the digital audio file is played at a secondclock frequency that is higher than the first clock frequency,increasing playback speed of the digital audio file.

In one embodiment, the steps of FIG. 3A are performed purely in thedigital domain. For example, although step 814 a may include convertingthe digital audio file into an analog signal, in one embodiment thedigital audio file is played purely in the digital domain. In step 814a, playback does not necessarily require the digital audio to be audibleto a listener. In a purely digital context, the digital audio file maybe played by processing it as if it were being converted into analog butwithout using a physical digital-to-analog converter. Instead, theprocessor may route the audio information to a plugin, which utilizesthe second clock frequency. A processor may read the digital audio fileat the second playback frequency, and apply formulas (representinganalog models) to the resulting time-based array of bits to applycompression, limiting, and/or equalization. Because accurate digitalmodels of the analog components may behave similarly to thecorresponding analog components, processing the digital audio file atthe second clock frequency as described herein may provide similarbenefits to those already outlined with regard to processing the analogsignal with the analog circuit.

The other steps of FIG. 3A may also be carried out in the digital domainin one embodiment. As discussed above, at step 316, the processor mayapply audio processing (e.g., via plugin) comprising at least one ofcompression, limiting, and equalization to the audio while utilizing(e.g., internally playing the audio file at) the second clock frequency.

Then, at step 320 a, the processor may create a modified digital audiofile based on the applied audio processing. This may include saving themanipulated digital audio file, which has modified contents, over thesource (i.e., original) digital audio file. In another embodiment, themanipulated digital audio file is saved separately from the originaldigital audio file.

Finally, at step 322 a, the processor may change the playback clockfrequency of the modified digital audio to the first clock frequency,ensuring proper playback of the manipulated digital audio file.

Unlike FIG. 3A, the steps of FIG. 3B necessarily require converting thedigital audio file for processing within the analog domain. Inparticular, step 814 b includes converting the digital audio into ananalog audio signal while the digital audio is playing at the highersecond clock frequency.

Then, step 316 b includes applying at least one of compression,limiting, and equalization to the analog audio signal. This can include,for example, passing the analog audio signal through a circuitcontaining analog hardware that modifies characteristics of the audiosignal.

At step 318, the an analog-to-digital converter may convert themanipulated analog audio signal into manipulated digital audio. Then, atstep 320 b, the processor stores the manipulated digital audio on acomputer-readable storage medium. This computer readable storage mediummay be located on a separate workstation than the workstation thatplayed the digital audio file in one embodiment.

Finally, at step 322 b, the processor may set metadata of themanipulated digital audio to indicate the first clock frequency fornormal playback speed.

FIG. 4 is an exemplary flow chart with a non-exhaustive listing of stepsthat may be performed by a digital audio workstation (DAW) and an audioprocessing device 205 while they interface with one another, inaccordance with an embodiment.

In this example, box 400 contains a non-exhaustive listing of stepsperformed by the DAW. Box 405 contains a non-exhaustive listing of stepsperformed by the audio processing device.

At step 410, audio may be recorded in the DAW and stored as a digitalaudio file.

In one embodiment, the DAW receives input from the user to perform step415, which includes exporting at least a portion of the digital audiofile to the audio processing device. For example, the user may select asingle track within the sequencing environment to export. Alternatively,the user may select just a portion of a single track to export. Stillfurther, the user may select a mixdown of an entire mix to export.

In the example of FIG. 4, the digital audio file is exported prior toincreasing the playback clock frequency of the digital audio file.However, in an alternate embodiment, the frequency is increased prior toexporting the digital audio file. In that alternate embodiment, the DAWmay output a representation of the audio at the second clock frequencyfor modification, wherein the modification includes at least one ofcompression, equalization, and limiting. This representation can beanalog in one embodiment, or digital in another embodiment.

At step 420, an input interface of the audio processing device 205accepts and stores the digital audio file. The input interface caninclude an input port, receiver circuitry, and the processor, which mayreceive data according to a protocol recognized by the DAW.

At step 425, the processor increases the playback clock frequency of thedigital audio to a higher second frequency. In this way, the audioprocessing device 205 may modify a digital audio file that normally hasa first clock frequency for playback to instead have a second clockfrequency that is higher than the first clock frequency. As mentionedabove, this step may instead occur on the DAW in one embodiment.

At step 430, a digital-to-analog converter converts the digital audiofile into analog audio while the digital audio file plays at the secondclock speed.

At step 435, the analog signal is passed through the analog modificationcircuit, which applies at least one of compression, limiting, andequalization to the analog audio signal.

Then, at step 440, the analog-to-digital converter converts the modifiedanalog audio into modified digital audio.

At step 445, that modified digital audio may be stored on acomputer-readable storage medium, after which the playback clockfrequency is changed to the first frequency. This storage medium may belocated on the audio processing device 205 in one embodiment. In anotherembodiment, the computer-readable medium is located externally to theaudio processing device.

At step 450, an output interface outputs the modified digital audio. Inone embodiment, step 450 is performed in unison with step 445.

At step 460, the DAW imports the digital audio file form the audioprocessing device.

At step 470, the DAW implements the modified (i.e., manipulated) digitalaudio, which can include adding the modified digital audio file to asequencer environment to replace or provide an alternative to theportion of the digital audio file initially exported to the audioprocessing device.

Turning to FIG. 5, an exemplary flow chart is presented with anon-exhaustive listing of steps that may be performed by a digital audioworkstation (DAW) 115.

At step 510, the DAW may modify a digital audio file having a firstclock frequency for playback to have a second clock frequency that ishigher than the first clock frequency.

At step 520, the DAW may output a representation of audio based on thesecond clock frequency for modification. For example, the representationmay be digital in one embodiment, including reading the digital audiofile at the second clock frequency. In another embodiment, therepresentation may be a representative audio signal, such as the audiosignal resulting from performing a digital-to-analog conversion of thedigital audio file by the DAW or an external converter in communicationwith the DAW.

The outputted representation may then be modified externally from theDAW. This modification may include at least one of compression,equalization, and limiting, as previously discussed herein.

At step 530, after the representation of the audio is externallymanipulated, the DAW may receive a modified representation of the audioat the second clock frequency. In one embodiment, the modifiedrepresentation is a digital audio file. In another embodiment, themodified representation is an audio signal (e.g., the conversion mayoccur on the DAW).

At step 540, the DAW may store the modified representation of the audioas a modified digital audio file. If the modified representation is ananalog signal, this step includes converting the analog signal into themodified digital audio file.

At step 550, the DAW may convert the clock frequency of the modifieddigital audio file to the first clock frequency for proper playback.

Turning now to FIG. 6, illustrated therein is another audio processingsystem 600 configured in accordance with one or more embodiments of thedisclosure. The audio processing system comprises a remote device, i.e.,a client device 601, which is in communication with a server complex 602across a network 603. In one embodiment, the network 603 is theInternet. The client device 601 can take any of a variety of forms,including computers, tablet computers, smartphones, and other devices.Illustrating by example, in one embodiment the client device 601comprises a tablet computer running a web browser to access a userinterface 610. Advantageously, in one or more embodiments the userinterface 610 of the audio processing system 600 allows a user tocontrol one or more analog signal processors 605,606 remotely, acrossthe network 603, to master a digital audio file 604 in the analogdomain.

The server complex 602 can include one or more of computers,workstations, servers, and/or other devices. In FIG. 6, the servercomplex 602 is shown has having only a single server 607 for brevity.However, one or more intermediate servers can be disposed between thesingle server 607 and the client device 601. Additionally, as will bedescribed in more detail with reference to FIG. 11 below, in otherembodiments one or more cloud-based devices can be disposed between theserver complex 602 and the client device 601. The single server 607 ofthe server complex 602 can even be a cloud-based device. Otherconfigurations of the server complex 602 will be obvious to those ofordinary skill in the art having the benefit of this disclosure.

In one or more embodiments, the server complex includes a first digitalaudio workstation 608 and a second digital audio workstation 609. Thefirst digital audio workstation 608 and the second digital audioworkstation 609 of this embodiment are disposed to either side of one ormore analog signal processors 605,606. The first digital audioworkstation 608, the one or more analog signal processors 605,606, andthe second digital audio workstation 609 work in tandem to carry outaudio processing. In this example, the components of the server complex602 are split into a digital domain, e.g., signals processed by eitherthe first digital audio workstation 608 and a second digital audioworkstation 609, and analog domain, e.g., signals processed by the oneor more analog signal processors 605,606.

The server complex 602 provides users with access to the masteringcomponents, e.g., the first digital audio workstation 608, the one ormore analog signal processors 605,606, and the second digital audioworkstation 609 across the network 603. In one embodiment, the servercomplex 602 delivers a user interface 610 for presentation on the clientdevice 601.

In one or more embodiments, the user interface 610 is operable toreceive one or more analog domain control settings 611 that are used tocontrol the one or more analog signal processors 605,606 and/or thefirst digital audio workstation 608 and the second digital audioworkstation 609. For instance, in this illustrative example the userinterface 610 comprises a loudness level selection 612 defining aloudness level 613 associated with a digital audio file 604.

In one or more embodiments, the user interface 610 is also operable toreceive the digital audio file 604. In this illustrative embodiment, theuser interface 610 also comprises a network upload portal 614 throughwhich the server complex 602 can receive an upload 615 of the digitalaudio file 604 through the network upload portal 614.

When a user wishes to master a digital audio file 604 in the analogdomain, they simply navigate to the user interface 610. For example, inone embodiment the user launches a web browser on the client device 601to navigate to the user interface 610. Once at the user interface 610,the user may enter identifying information, such as their name anddigital audio file title. The user can also upload 615 the digital audiofile 604 to the server complex 602 using the network upload portal 614.

In one or more embodiments, the user also establishes one or more analogdomain control settings 611 that will be used in the analog domainduring the mastering. In this simple example, the user selects aloudness level 613 associated with a digital audio file 604 using theloudness level selection 612. The options presented in this example are(1) Moderate loudness level, which is good for dynamic mixes andacoustic songs, (2) Loud loudness level, which is suitable for standardmixes with moderate compression, (3) Louder loudness level, which issuitable for well balanced mixes that are well compressed, and (4) aLoud/Safe loudness level, which is suitable for ballads and softer songsthat need to be louder. In this illustrative embodiment, the loudloudness level is the default value, and also happens to be the selectedloudness level 616 of the user. When the digital audio file 604 isuploaded, so too are the one or more analog domain control settings 611.

The server complex 602 then receives the digital audio file 604 and theone or more analog domain control settings 611 from the client device601 across the network 603. In this illustrative example, the servercomplex 602 includes a first digital audio workstation 608 that storesthe received digital audio file 604, and optionally the one or moreanalog domain control settings 611. Alternatively, the server 607 storesthe one or more analog domain control settings. The digital audioworkstation 608 can comprise at least one processor and a memory orother computer readable storage medium. In one embodiment, the digitalaudio workstation 608 is a stand-alone device built specifically forhandling audio production, mixing, and/or processing. For example, thedigital audio workstation 608 may have an integrated mixer, audiosequencer, and/or effects capabilities. In another embodiment, thedigital audio workstation 608 can comprise a personal computer withsoftware being executed by a processor for the purpose of audioproduction, recording, mixing, and/or mastering.

The digital audio file 604 can be received by the server complex 602 inany of a variety of formats, including .WAV, .AIFF, MP3, SDII, AC3, DSD,or other audio file formats. For example, the digital audio file 604shown in FIG. 1 is a .WAV file, which is compatible with the Windows™operating system and typically contains non-compressed audioinformation. This means that the digital audio file 604 can be arelatively large file that contains all recorded audio information.However, other file types are possible. For example, the digital audiofile 604 can even include a video file type, such as .AVI, to the extentthat the video file type includes an audio track or portion.

The digital audio file 604 may also contain metadata 617 that specifiescharacteristics of the digital audio file 604, such as the bit rate andthe sample rate. Other characteristics can also be identified in themetadata 617 as well. For example, .WAV files contain a header that canindicate surround sound and speaker positions, provide informationregarding sample types, and supports defining custom extensions to theformat chunk. The sample rate may indicate the number of samples persecond used in a digital representation of an analog signal. The bitrate may indicate the number of bits used to represent the level of thesample. In theory, the higher the sample rate and bit rate, the closer adiscrete digital audio file represents the continuous analog audiosignal that it emulates.

In one or more embodiments, the first digital audio workstation 608comprises a digital-to-analog converter 618. The digital-to-analogconverter 618 can be configured as a sound card of the first digitalaudio workstation 608 in one embodiment. In another embodiment, thedigital-to-analog converter 618 can be configured as a standalonecomponent located externally to, but operable with, the first digitalaudio workstation 608.

The digital-to-analog converter 618 can receive the digital audio file604 from the server 607 of the server complex 602 and can convert thedigital audio file 604 to an analog signal 619. In this example, thefirst digital audio workstation 608 may convert the digital audio file604 into an analog signal 619 and send the analog signal 619 to the oneor more analog signal processors 605,606 for processing. It isunderstood that this conversion can utilize an external converter in oneembodiment.

In one or more embodiments, a control device 620 is operable to controlthe one or more analog signal processors 605,606. As will be describedin subsequent figures below, the control device 620 can take a varietyof forms, including one or more of a robotic arm, digitally controlledrelays, voltage controlled filters, digitally controlled servo-drivenpotentiometers, digitally controlled servo-driven attenuators, digitallycontrolled voltage controlled amplifiers, digitally controlled variablegain amplifiers, or combinations thereof. Other control devices suitablefor controlling the one or more analog signal processors 605,606 will beobvious to those of ordinary skill in the art having the benefit of thisdisclosure.

The control device 620 receives the one or more analog domain controlsettings 611 from the server 607 of the server complex 602 or the firstdigital audio workstation 608, and applies them to the one or moreanalog signal processors 605,606. Illustrating by example, if the one ormore analog signal processors 605,606 comprise resistors, capacitors,inductors, operational amplifier, vacuum tubes, transistors, or otheranalog components, as controlled by one or more mixing consoles 621,622,and the control device 620 is a robotic arm, the robotic arm can adjustthe knobs, sliders, and other controls of the mixing consoles 621,622 toapply setting adjustments to adjust the same to configure the one ormore analog signal processors 605,606 in accordance with the one or moreanalog domain control settings 611 received from the client device 601.

In one or more embodiments, once the digital audio file 604 is convertedto the analog signal 619, it is delivered from the first digital audioworkstation 608 to the one or more analog signal processors 605,606. Theone or more analog signal processors 605,606 receive the analog signal619 from the digital-to-analog converter 618 and apply at least onedynamic analog modification 681 to the analog signal 619 in accordancewith the one or more analog domain control settings 611 received fromthe client device 601 (since the control device 620 has set the mixingconsoles 621,622 in accordance with the one or more analog domaincontrol settings 611 to the analog signal 619 to obtain modified analogsignal 623.

The dynamic modifications or effects applied by the one or more analogsignal processors 605,606 may include at least one of compression,limiting, and equalization. In some embodiments, additional effects arepossible, such as stereo field effects, exciter effects, tape emulationeffects, etc. In order to apply these effects, one or more analog signalprocessors 605,606 one or more hardware modules. (Examples of suchhardware modules, integrated into the analog signal processors 605,606of FIG. 6, are shown in FIG. 7 as modules 721,722,723.) The modules maycomprise any known combination of circuitry and analog components forapplying compression, limiting, and/or equalization, depending on thedynamic effect applied by the particular module.

Additionally, each of the modules may be connected to one another in aneffects chain in one embodiment. In an effects chain, the output fromone module can serve as an input for another module. For example, acompressor module (721) may output a modified analog signal that isreceived as an input at a limiter module (722). The output of limitermodule (722) may then be received as an input of equalization module(723).

The compressor module (721) is used to compress the dynamic range of theanalog signal 619. This type of compression is distinct from datacompression, in which the information is optimized for a smaller filesize. In dynamic range compression, quiet sounds can be made louder byreducing the dynamic range of loudness and amplifying the quiet sounds.

The type of compression applied may vary between embodiments. Forexample, a peak sensing compressor may respond to an instantaneous levelof the input signal. This type of compression may provide tighter peakcontrol, but can yield very quick changes in gain reduction, which undertraditional audio processing methods can lead to audible distortion.Alternatively, an averaging compressor may be used to apply an averagingfunction (such as root mean squared (“RMS”)) on the input signal beforeits level is compared to the threshold. Some compressors may includecontrols or inputs to set a compression ratio, which typicallydetermines the reduction of signal loudness, and a gain level toincrease the loudness of the audio signal. Other controls, such asattack, release, and knee control may be provided to help shape thecompression. The attack may determine the period when the compressordecreases gain to reach the level governed by the ratio. The release maydetermine the period when the compressor is increasing gain to the levelgoverned by the ratio, or, to zero dB, once the level has fallen belowthe threshold. The length of each period may be determined by the rateof change and the required change in gain. In one embodiment, the attackand release times are adjustable by the user. In another embodiment, theattack and release times determined by the circuit design and cannot beadjusted by the user.

A limiter module (722) may receive a modified analog signal fromcompressor module (721). Limiting, as provided by the limiter module(722), is technically another form of compression that includes a veryhigh compression ratio. For example, a compression ratio between 60:1and .infin.:1 may be used in limiting. The purpose of limiting isgenerally to keep the analog signal 619 level below 0 dB, to avoid“clipping.” Audio engineers and producers typically try to avoidclipping because clipping results in a harsh and typically undesirableaudio artifact. In an alternate embodiment, limiting is not appliedbecause the converters effectively limit the audio signal when thelow-frequency information is no longer present.

With prior systems, if limiting is relied on too heavily to reduce audiolevels, overload and distortion can occur. For example, when the signalprocessed by the limiter is consistently far above 0 dB, the amount ofcompression applied by the limiter can cause distortion for similarreasons as explained above with regard to compressors. Advantageously,by being able to remotely control the one or more analog signalprocessors 605,606 remotely with the one or more analog domain controlsettings 611, this overload and distortion can be avoided without theneed of traveling to the studio.

An equalization module (723) may apply equalization to the analog signal619. Equalization may alter the frequency response of the audio signal,amplifying some frequencies and/or reducing some frequencies. This canbe used, for example, to emphasize different frequencies across thestereo field to make particular sounds, instruments, and/or voices standout in an audio mix. However, analog equalization hardware, particularlycheap equalization hardware commonly found in home studios, canintroduce distortion in the low frequencies if the audio signal is tooloud for the equalizer to handle. Advantageously, by being able toremotely control the one or more analog signal processors 605,606remotely with the one or more analog domain control settings 611, thisoverload and distortion can be avoided without the need of traveling tothe studio. Note that while in this example compression is providedfirst, then limiting, and then equalization, effects may also beprovided in other orders. For example, equalization may be appliedbefore any compression in another embodiment.

In this illustrative embodiment, the server complex 602 also includes asecond digital audio workstation 609. In one embodiment, the seconddigital audio workstation 609 comprises an analog-to-digital converter624. After the one or more analog signal processors 605,606 apply thedynamic analog modification(s) to the analog signal 619 in accordancewith the one or more analog domain control settings 611 received fromthe client device 601 across the network 603 to create the modifiedanalog signal 623, the analog-to-digital converter 624 converts themodified analog signal 623 into a modified digital audio file 625.

In one or more embodiments, the modified digital audio file 625 is thenstored in a memory device or other non-transitory computer-readablestorage medium at the second digital audio workstation 609. Examples ofsuch a memory or non-transitory computer-readable storage medium includea disk drive or some other storage medium. However, in anotherembodiment, the memory or other non-transitory computer-readable storagemedium is located on a separate product or workstation from the seconddigital audio workstation 609, either in the cloud or within the servercomplex 602. One example of such a separate device would be server 607.

In one or more embodiments, the server complex 602 presents a networkdownload portal 626 facilitating download of the modified digital audiofile 625 across the network 603. Using the network download portal 626,the user can download the modified digital audio file 625 to the clientdevice 601 or another storage medium. Accordingly without ever visitingthe server complex 602 or its associated studio, the user is able tomaster the digital audio file 604, using their own prescribed settingsfor the one or more analog signal processors 605,606. Prior art systemsare simply unable to allow such remote control of analog components.

In one or more embodiments, the server complex 602 includes componentsthat synchronize delivery of the analog signal 619 to the one or moreanalog signal processors 605,606 and receipt of the modified analogsignal 623 from the one or more analog signal processors 605,606. Onereason such synchronization may be required is due to the fact thatdigital files are non-transitory, while audio signals are transitory.Accordingly, when an audio signal is delivered to the one or more analogsignal processors 605,606, it passes therethrough without storage. Wherea first digital audio workstation 608 and a second digital audioworkstation 609 are included, synchronization may be required to alert,for example, the second digital audio workstation 609 that the firstdigital audio workstation 608 is delivering the analog signal 619 to theone or more analog signal processors 605,606 so that the second digitalaudio workstation 609 can listen to receive the modified analog signal623.

In this illustrative embodiment, the server 607 of the server complex602 is operable to synchronize delivery of the analog signal 619 to theone or more analog signal processors 605,606 and the conversion of theanalog signal 619 to the modified digital audio file 625. First digitalaudio workstation 608 is referred to as the “pitch” workstation, whilethe second digital audio workstation 609 is referred to as the “catch”workstation. This is the convention because “pitch” pitches the analogsignal 619 to the one or more analog signal processors 605,606, while“catch” catches the modified analog signal 623 from the one or moreanalog signal processors 605,606.

In one or more embodiments, server 607 synchronizes pitch and catch byactuating the analog-to-digital converter 624 of catch when thedigital-to-analog converter 618 of pitch delivers the analog signal 619to the one or more analog signal processors 605,606. This allows catchto receive the modified analog signal 623 as it is output from the oneor more analog signal processors 605,606. Thus, in one or moreembodiments the server complex 602 causes a concurrent initiation of aconversion of the digital audio file 604 to the analog signal 619 at thefirst digital audio workstation 608 to deliver the analog signal 619 tothe one or more analog signal processors 605,606 and conversion of themodified analog signal 623 at the second digital audio workstation 609to a modified digital audio file 625 after the application of the atleast one dynamic analog modification 681.

In one or more embodiments, prior to applying the dynamic analogmodification 681 to the analog signal 619, the sample rate of thedigital audio file 604 can be converted to a predefined or standardsample rate. The sample rate may indicate the number of samples persecond used in a digital representation of an analog signal. The bitrate may indicate the number of bits used to represent the level of thesample. In theory, the higher the sample rate and bit rate, the closer adiscrete digital audio file represents the continuous analog audiosignal that it emulates.

It is known that the normal playback or recording frequency, referred toas a “sample rate” in the digital domain, can vary between differentdigital audio files. The playback frequency can be identified in themetadata 617 as the sample rate associated with the digital audio file604. For example, the standard sample rate, i.e., normal playbackfrequency, used for digital audio files on audio compact discs is 44,100samples per second (44.1 kHz), with 16 bits of information per sample.Digital Video Discs on the other hand, contain digital audio files witha sample rate of 48 kHz and 24 bits of information per sample.

Audio and music files can be recorded at a variety of different samplerates. Each different sample rate results in a different normal playbackfrequency. For example, some professional audio hardware provides theoption for sample rates of 88.2 kHz, 96 kHz, and/or 192 kHz. Even thoughstandard audio gear and related applications tend to call for digitalaudio files with 44.1 kHz or 48 kHz sample rates, higher sample ratescan be useful in audio recording applications where effects are appliedto ensure that the modified source information is as close to theoriginal analog signal, e.g., the signal generated by pressure on amicrophone diaphragm, as possible.

Accordingly, in one or more embodiments the first digital audioworkstation 608 will adjust the effective sample rate of the digitalaudio file to a predefined rate by changing the metadata. This isdifferent from resampling. By changing the metadata, audio equipmentprocessing the file believes the file to be recorded at one sample rate,i.e., by reading the metadata, when the file was actually recorded at adifferent sample rate. Accordingly, when the equipment processes thatfile, the audio characteristics will change from their original form.For example, the song might sound faster and at a higher pitch, oralternatively slower and at a lower pitch.

Illustrating example, if the digital audio file 604 has associatedtherewith a sample rate of 192 kHz, as indicated by the metadata, and asample rate of 44.1 kHz is desired, the metadata can be changed to makeequipment believe that the digital audio file 604 was recorded at 44.1kHz and not 192 kHz. When the equipment processes the digital audio file604, it will therefore sound slower and lower. Thus, the effectivesample rate can be adjusted to this lower rate by altering the metadataassociated with the file, which alters the speed of playback. Similarly,if the digital audio file 604 has a sample rate of 88.2 kHz, and asample rate of 192 kHz is desired, the first digital audio workstation608 can convert the effective sample rate to the higher rate by changingthe metadata in similar fashion. Adjusting metadata 617 is preferable toresampling at a different sample rate because the latter generally canlead to fidelity loss and are avoided by audio professionals ifpossible.

Additionally, in other embodiments, prior to applying the dynamic analogmodification 681 to the analog signal 619, the first digital audioworkstation 608 can change the clock frequency associated with thedigital audio file 604. Illustrating by example, the digital audio file11 may have metadata 617 indicating a clock frequency that should beused for normal playback to preserve the proper frequencycharacteristics of the music represented by the digital audio file 604.For instance, the metadata 617 may indicate a sample rate of 44.1 kHz.In one embodiment, the sample rate may also be the clock frequency thatshould be used for normal playback to preserve the proper frequencycharacteristics of the music represented by the digital audio file 604.

Even where the metadata 617 does not indicate the clock frequency, thesample rate can be extrapolated into a clock frequency to use for normalplayback. For instance, because each sample of the digital audio file604 contains multiple bits-worth of information, if the audio processingsystem 100 ties the clock to a particular amount of data to beprocessed, the actual clock frequency for playback may also depend onthe bit rate, which also may be indicated by metadata 617 in the digitalaudio file 604. However, the sample rate indicated by metadata 617 inmany systems indicates the actual clock frequency for normal playback,eliminating the need for the processor to calculate a different clockfrequency for use in playback. However, either embodiment is consideredto indicate a first clock frequency for a processor to use for normalplayback.

In one or more embodiments, when the first digital audio workstation 608begins to convert the digital audio file 604 with the digital-to-analogconverter 618, it does so by routing informational chunks of the digitalaudio file 604 at a specified playback clock frequency, which may or maynot be equal to the sample rate. The digital-to-analog converter thenconverts the informational chunks into an analog signal 619.

In one or more embodiments, the first digital audio workstation 608 maycause the digital audio file 604 to play at a second clock frequencythat is different from the first clock frequency that should be used fornormal playback to preserve the proper frequency characteristics of themusic represented by the digital audio file 604. The second clockfrequency can be higher or lower than the first clock frequency.However, in one or more embodiments the second clock frequency isgreater than the first clock frequency. Illustrating by example, in oneor more embodiments the first digital audio workstation 608 may set themetadata 617 of the digital audio file 604 to indicate a second clockfrequency for playback that is double the first clock frequency.However, other combinations are possible, such as a 25 percent higherclock frequency.

Using the second clock frequency for playback that is higher than thenormal clock frequency causes the digital audio file 604 to playback ata faster speed than normal. As a result, the digital audio file 604exhibits higher frequency characteristics than when played at the normalplayback frequency. It also completes playback more quickly. By doublingthe clock frequency, a digital audio file with audio information up to22,500 Hz can have audio information up to 44,100 KHz, which is faroutside the range of human hearing.

In one embodiment, the second clock frequency is chosen to substantiallyreduce or virtually eliminate audio frequency information below 250 Hz.This low frequency information often creates a “muddy” sound and may bethe cause of distortion created by digital-to-analog converters and/oranalog components, such as a compressor, or digital components, such asdigital audio processors. The exact clock frequency needed to raise thelow frequency information above this threshold may vary depending on thesource audio information. For example, if an audio file has substantialaudio information at 200 Hz, a 25 percent increase in clock frequencywill move that audio information to above 250 Hz. In one embodiment, theideal clock frequency is chosen automatically by the processor, whichanalyzes the digital audio file to determine which clock frequency willmove audible levels of audio information to above 100 Hz.

In one embodiment, the digital-to-analog converter 618 converts thedigital audio into an analog audio signal while the digital audio isplaying at the higher second clock frequency. Because the first digitalaudio workstation plays the digital audio file 604 at the higher secondclock frequency, less low frequency information is passed through theconverters. This reduces distortion and allows for a louder analog audiosignal. Eliminating and/or reducing low frequency information alsolightens the load on these components, including the analog input of theanalog-to-digital converter 624, in which low-frequency information canaccount for significant portions of current, causing overloading and/ordistortion. This results in a clearer analog signal 619.

The analog signal 619, as a result, may require less compression sinceit is already louder. This, in turn, also allows for maintainingdynamics in volume while still achieving commercial loudness levels.This further leads to more clarity in the digital-to-analog conversion,since lower frequencies are often the cause of the most audibledistortion during the conversion process.

The clock signal used to play the digital audio file 604 at the highersecond clock frequency may be generated by the first digital audioworkstation 608 in one embodiment. Alternatively, a modulecommunicatively coupled to the digital audio file 604 may be responsiblefor generating the clock signal in another embodiment. For example, aseparate clock module may be used to reduce an effect called jitter byhaving the clock module supply the digital audio workstation 608 with amore accurate clock signal. Other modules, such as the digital-to-analogconverter 618, may alternatively supply the clock signal to theprocessor.

Although the sample rate can be changed in metadata 617 to reflect thesecond clock frequency in one embodiment, an alternate embodiment doesnot alter the metadata 617. Instead, the digital audio workstation 608may notify an external converter of the playback clock frequency to use.Or the user may select the clock frequency using the user interface 610.The external converter may not check the metadata 617 of the digitalaudio file, but instead will supply the clock at the frequency indicatedby the digital audio workstation 608 or user. In this embodiment, afterthe audio has been processed, the resulting modified digital audio filemay already contain the correct metadata 617 for sample rate. However,in one embodiment, the external converter must be notified to change theclock frequency back to the first frequency for normal playback.

Where changed playback frequencies are employed, once the modifiedanalog signal 623 is output from the last effects module, the modifiedanalog signal 623 is converted back into a modified digital audio file625 through use of an analog-to-digital converter 624. In one or moreembodiments, this conversion occurs without changing the speed of themodified digital audio file 625. In other words, the converted fileinitially may be set to play at the second clock frequency.

In an alternate embodiment, the analog-to-digital converter 624 may beset to change the playback clock frequency of the modified analog signal623 as compared to the original digital audio file 604 without modifyingthe metadata 617. In this instance, the playback clock frequencysupplied, e.g., using a crystal oscillator, by the analog-to-digitalconverter 624 may be changed accordingly to cause the modified digitalaudio file 625 to play at the same speed as the original digital audiofile 604 with the second playback frequency. In one such embodiment, theexternal converter may not know the contents of the metadata 617 at anypoint in the process. In this way, no changes to the sample ratespecified in metadata 617 occur in one embodiment.

Once the second digital audio file has been stored at the second digitalaudio workstation 609, or alternatively in the server 607, in oneembodiment the metadata 617 of the modified digital audio file 625 canbe changed to indicate the first clock frequency for normal playbackspeed. This effectively restores the frequency response of the modifieddigital audio file 625 heard when played, eliminating any “chipmunkeffect” caused by setting the playback frequency to the second frequencyprior to dynamic enhancement.

Turning now to FIG. 7, illustrated therein is an alternate audioprocessing system 700 configured in accordance with one or moreembodiments of the disclosure. The audio processing system 700 of FIG. 6is operable with a client device (601), just as was the audio processingsystem (600) of FIG. 6. The client device (601) can be in communicationwith a server complex 702 across a network (603). Here, the servercomplex 702 is shown separated into a digital domain 741 and an analogdomain 742. As before, the client device (601) accesses a user interface(610) of the audio processing system 700, which allows a user to controlone or more analog signal processors 705 remotely, across the network(603), to master a digital audio file 704 in the analog domain.

In this illustrative embodiment, the server complex 702 has a singledigital audio workstation 708 that functions both as an audioworkstation and a server. The digital audio workstation 708 is operablewith the one or more analog signal processors 705. The server complex702 provides users with access to the mastering components, e.g., thedigital audio workstation 708 and the one or more analog signalprocessors 705 across the network (603). As before, the user interface(610) is operable to receive one or more analog domain control settings711 that are used to control the one or more analog signal processors705 and/or the digital audio workstation 708.

The server complex 702 then receives the digital audio file 704 and theone or more analog domain control settings 711 from the client device(601) across the network (603). The digital audio file 704 may alsocontain metadata (617) as previously described.

The digital audio workstation 708 comprises a digital-to-analogconverter 718 and an analog-to-digital converter 724. Thedigital-to-analog converter 718 receives the digital audio file 704 andconverts the digital audio file 704 to an analog signal 719. Thedigital-to-analog converter 718 then sends the analog signal 719 to theone or more analog signal processors 705,606 for processing.

A control device 720 is operable to control the one or more analogsignal processors 705. The control device 720 receives the one or moreanalog domain control settings 711 from the digital audio workstation708 and applies them to the one or more analog signal processors 705.The control device can adjust the knobs, sliders, and other controls ofthe mixing consoles to apply setting adjustments to configure the one ormore analog signal processors 705 in accordance with the one or moreanalog domain control settings 711.

The one or more analog signal processors 705, receive the analog signal719 from the digital-to-analog converter 718 and apply at least onedynamic analog modification to the analog signal 719 in accordance withthe one or more analog domain control settings 711 to obtain modifiedanalog signal 722. The dynamic modifications or effects applied by theone or more analog signal processors 705 may include at least one ofcompression, limiting, equalization, or combinations thereof.

One or more hardware modules 721,722,723 apply the compression,limiting, equalization, or combinations thereof. A compressor module 721may output a modified analog signal that is received as an input at alimiter module 722. The output of limiter module 722 may then bereceived as an input of equalization module 723.

The compressor module 721 is used to compress the dynamic range of theanalog signal 719. The limiter module 722 may receive a modified analogsignal from compressor module 721. The equalization module 723 thenapplies equalization to the analog signal 719.

After the one or more analog signal processors 705,606 apply the dynamicanalog modification(s) to the analog signal 719 in accordance with theone or more analog domain control settings 711, the analog-to-digitalconverter 724 converts the modified analog signal 723 into a modifieddigital audio file as previously described. The server complex 702 canpresent a network download portal (626) that allows a user to downloadthe modified digital audio file across the network (603). Accordinglywithout ever visiting the server complex 702 or its associated studio,the user is able to master the digital audio file 704, using their ownprescribed settings for the one or more analog signal processors 705.Prior art systems are simply unable to allow such remote control ofanalog components.

Turning now to FIG. 8, illustrated therein is a system level diagram ofone explanatory audio processing system 800 configured in accordancewith one or more embodiments of the disclosure. As shown in FIG. 8, aremote electronic device 801 is in communication with a server complex802 across a network 803. The remote electronic device 801 can beconsidered to be an “off host” computer in that it is separated from thehost, i.e., the server complex 802 in this embodiment, by the network803. In one or more embodiments, the network 803 comprises the Internet.However, other networks could be substituted for the Internet. Forexample, the network 803 can comprise a wide area network, local areanetwork, ad hoc network, or other network. Still other networks will beobvious to those of ordinary skill in the art having the benefit of thisdisclosure.

At step 804, the remote electronic device 801 navigates to a userinterface, which in one embodiment is a web portal. In one or moreembodiments, the user interface includes a network upload portalreceiving an upload of the digital audio file through the network uploadportal At step 805, the remote electronic device 801 uses the userinterface to select one or more analog domain control settings and toattach a digital audio file. The analog domain control settings could beselected via a preset selector. For instance, in the embodiment of FIG.6 above one or more analog domain control settings were selected via aplurality of loudness indicators, each of which corresponds to one ormore preset levels and controls of analog signal processing equipment.

The one or more analog domain control settings could be selected inother ways as well. For example, as will be described below withreference to FIG. 11, sliders or other level adjusters can be used toselect the one or more analog domain control settings. As will bedescribed with reference to FIG. 12, the one or more analog domaincontrol settings could be created using a virtual representation of thestudio, including a mixing console or other representation of the analogsignal processors. Still other techniques for entering the analog domaincontrol settings will be obvious to those of ordinary skill in the arthaving the benefit of this disclosure. In one or more embodiments, theone or more analog domain control settings created at step 805 areconverted to a format convenient for transmission across the network803, such as via an Extensible Markup Language (XML) file.

At step 806, the remote electronic device 801 transmits the digitalaudio file and the one or more analog domain control settings across thenetwork 803 to the server complex 802. In one or more embodiments, thedigital audio file and/or the one or more analog domain control settingsmay be transmitted and/or stored on one or more intermediate serversduring this process. For example, one or more “cloud” servers may storethe digital audio file and/or the one or more analog domain controlsettings so that they are accessible by both the server complex 802 andthe remote electronic device 801.

At step 807, the serve complex receives the digital audio file and theone or more analog domain control settings from the remote electronicdevice 801 across the network 803. At step 808, a digital-to-analogconverter at the server complex 802 receives the digital audio file andconverts the digital audio file to an analog signal. At step 809, acontrol device operable with one or more analog signal processors at theserver complex 802 applies setting adjustments to the one or more analogsignal processors in accordance with the one or more analog domaincontrol settings received from the remote electronic device 801.Examples of control devices include robotic arms, digitally controlledrelays, voltage controlled filters, digitally controlled servo-drivenpotentiometers, digitally controlled servo-driven attenuators, digitallycontrolled voltage controlled amplifiers, or digitally controlledvariable gain amplifiers. Still other techniques for mechanicallycontrolling the potentiometers, sliders, levers, and other controls of amixing console or other control mechanism for analog signal processingcircuits will be obvious to those of ordinary skill in the art.

At step 810, the one or more analog signal processors receive the analogsignal from the digital-to-analog converter and apply at least onedynamic analog modification to the analog signal. In one or moreembodiments, this modification occurs in accordance with the one or moreanalog domain control settings received from the remote electronicdevice 801 due to the control device applying the setting adjustments ofthe previous paragraph. When the analog signal processors apply the oneor more dynamic analog modifications to the analog signal, this createsa modified analog audio signal. The dynamic analog modifications caninclude one or more of compression, limiting, or equalization. Otherdynamic analog modifications will be obvious to those of ordinary skillin the art having the benefit of this disclosure.

At optional step 811, where the server complex 802 includes multipledigital audio workstations, the server complex 802 can optionallysynchronize delivery of the analog signal to the one or more analogprocessors and the conversion of the analog signal to the modifieddigital audio file. At step 812, an analog-to-digital converter of theserver complex 802 converts the modified analog audio signal to amodified digital audio file. This optional step 812 can includeinitiation of a conversion of the digital audio file to the analogsignal at the first digital audio workstation to deliver the analogsignal to the one or more analog signal processors and conversion of theanalog signal at the second digital audio workstation to a seconddigital audio file after the application of the at least one dynamicanalog modification.

In one or more embodiments, the server complex 802 can make electronicaudio signals available to the remote electronic device 801 during anyof steps 807-812. Accordingly, a user at the remote electronic device801 can listen to the manipulation of the audio corresponding to any ofthe digital audio file, the analog audio signal, and/or the modifieddigital audio file in real time. Said differently, a user at the remoteelectronic device 801 can aurally monitor the mastering process, as itoccurs, in real time.

At step 813, the server complex 802 makes the modified digital audiofile available to the remote electronic device 801. This can be done inany of a number of ways. In the simplest embodiment, the server complex802 simply transmits the modified digital audio device to the remoteelectronic device 801 across the network 803. In another embodiment, theserver complex 802 delivers the modified digital analog file to a cloudserver that is accessible by the remote electronic device 801 and fromwhich the remote electronic device 801 can download the modified digitalaudio file.

In still another embodiment, the server complex 802 makes a networkdownload portal available to the remote electronic device 801. Thenetwork download portal facilitates download of the modified digitalaudio file. For example, in one embodiment the server complex 802 canemail a link to the remote electronic device 801. Clicking on the linkinitiates download of the modified digital audio file to the remoteelectronic device 801 from the server complex. Still other mechanismsfor making the modified digital audio file available to the remotecomputer will be obvious to those of ordinary skill in the art havingthe benefit of this disclosure.

At step 814, the remote electronic device 801 receives the modifieddigital audio file. Using the audio processing system 800, the remoteelectronic device 801 has been able to remotely control analog signalprocessing circuitry to perform true, high fidelity, analog mastering onan uploaded digital audio file.

Turning now to FIGS. 9 and 10, illustrated therein are different controldevices 620 suitable for use with one or more embodiments of thedisclosure. Beginning with FIG. 9, illustrated therein is a servercomplex 902 operable with one or more analog signal processors 905,906being controlled by mixing consoles 921,922. The mixing consoles921,922, also known as mixing boards, the boards, the dang boards, thesound boards, or the audio mixers, allow knobs, sliders, switches, andother potentiometer-like devices to control signal levels, frequencycontent, dynamics, and other effects of the one or more analog signalprocessors 905,906. A mastering engineer adjusts these controls withtheir fingers while listening critically to the audio signals. However,with embodiments of the disclosure, a remote device controls the samewith analog domain control settings.

The question thus becomes how to control mechanical devices with digitalinformation. In this illustrative embodiment, a robotic arm 950 receivesthe analog domain control settings 411 and makes the necessaryadjustments to the mixing consoles 921,922. Thus, a user can control themixing consoles 921,922 remotely with just a few clicks of the mouse.Advantageously, they can become mastering engineers taking advantage ofthe expensive analog equipment of a mastering studio without having totravel to the studio. The robotic arm 950 essentially becomes a remoteextension of their arm to make analog masters.

A robotic arm 950, while extremely effective, is not the only way tocontrol the controls of the mixing consoles 921,922. Turning now to FIG.10, illustrated therein are various other ways to control the knobs1051, sliders 1052, and switches 1053 of a mixing console 1021.

In one embodiment, the control device 620 comprises one or moredigitally controlled relays 1054. In another embodiment, the controldevice 620 can comprise one or more voltage-controlled filters 1055. Inyet another embodiment, the control device 620 comprises one or moredigitally controlled servo-driven potentiometers 1056. In still anotherembodiment, the control device 620 comprises one or more digitallycontrolled servo-driven attenuators 1057. In still another embodiment,the control device 620 comprises one or more digitally controlledvoltage controlled amplifiers 1058. In yet another embodiment, thecontrol device 620 comprises one or more digitally controlled variablegain amplifiers 1059. Each of these devices is able to receive a digitalinput and convert that digital input into analog, mechanical action,e.g., turning knobs 1051, translating sliders 1052, toggling switches1053, and making other adjustments to the mixing console 1021. Moreover,these various control devices can be used in combination as well. Stillother control devices will be obvious to those of ordinary skill in theart having the benefit of this disclosure.

Turning now to FIG. 11, illustrated therein is another audio processingsystem 1100 configured in accordance with one or more embodiments of thedisclosure. The audio processing system 1100 comprises a client device1101 in communication with a server complex 1102 across a network 1103.In this illustrative embodiment, the network 1103 comprises “the cloud.”

The cloud, which represents one or more networked servers 1160 ornetwork-based service providers 1161, can provide one or more computingservices that are available to one or both of the client device 1101 orthe server complex 1102. Illustrating by example, the one or morenetworked servers 1160 can include a collection of computing devices,which can be located centrally or distributed, that provide cloud-basedservices to one or both of the client device 1101 or the server complex1102 via a network 1103 such as the Internet. The cloud can be used, forinstance, by the server complex 1102 or the client device 1101 tooffload various computing tasks such as processing user input,presenting user interfaces, storing data, and so forth.

The cloud can, for example, provide services such as the presentation ofthe user interface 1110, the provision of the network upload portal1114, the client download portal 1181, storage of the digital audio file1104, the modified digital audio file 1125, or the one or more analogdomain control settings 1111. Service providers 1161 can customize thescreen size, display capability, file storage options, messaging,communications between the client device 1101 and the server complex1102, or other services. Still other uses for the cloud and itscorresponding networked servers 1160 or services providers 1161 will beobvious to those of ordinary skill in the art having the benefit of thisdisclosure.

The client device 1101 initially accesses the user interface 1110. Inone or more embodiments, the user interface 1110 is operable to receiveone or more analog domain control settings 1111 that are used to controlthe one or more analog signal processors 1105,1106 and/or the firstdigital audio workstation 1108 and (where included) the second digitalaudio workstation 1109.

In this illustrative embodiment, the one or more analog domain controlsettings 1111 comprise a plurality of sliders by which the desiredamount of compression 1163, equalization 1164, limiting 1165, or otherparameters can be controlled. Additionally, adjusters for clockfrequency 1166 can be included to clock the digital audio file 1104 at ahigher frequency while applying analog signal modifications aspreviously described can be provided as well.

In one or more embodiments, the user interface 1110 is also operable toreceive the digital audio file 1104. In this illustrative embodiment,the user interface 1110 also comprises a network upload portal 1114through which the server complex 1102 can receive an upload of thedigital audio file 1104 through the network upload portal 1114. In oneor more embodiments, the digital audio file 1104 is stored on anetworked server 1160 in the cloud so that it is accessible by both theclient device 1101 and the server complex 1102.

The server complex 1102 downloads the digital audio file 1104 and theone or more analog domain control settings 1111 form the networkedservers 1160 of the cloud. A digital-to-analog converter 1118 convertsthe digital audio file 1104 to an analog signal 1119. A control device1120 receives the one or more analog domain control settings v11 andapplies them to the one or more analog signal processors 1105,1106. Oncethe digital audio file 1104 is converted to the analog signal 1119, theone or more analog signal processors 1105,1106 apply at least onedynamic analog modification 1121 to the analog signal 1119 in accordancewith the one or more analog domain control settings 1111. The dynamicmodifications or effects applied by the one or more analog signalprocessors 1105,1106 may include at least one of compression, limiting,and equalization.

After the one or more analog signal processors 1105,1106 apply thedynamic analog modification(s) to the analog signal 1119 in accordancewith the one or more analog domain control settings 1111, ananalog-to-digital converter 1124 converts the modified analog signal1123 into a modified digital audio file 1125. In one embodiment, theserver complex 1102 then uploads the modified digital audio file 1125 toone of the networked servers 1160 in the cloud for retrieval by theclient device 1101.

While two explanatory user interfaces have been described above withreference to FIGS. 6 and 11, it should be noted that numerous others canbe used in accordance with one or more embodiments of the disclosure.For example, turning to FIG. 12, in this embodiment another userinterface 1210 is being presented on a tablet computer 1270. In thisillustrative embodiment, the user interface 1210 comprises a virtualpresentation of the mixing consoles 1221 used to control the analogsignal processors 1205. Accordingly, a user 1271 can pan across themixing console 1221 with their finger 1272 making adjustments to thevarious knobs, switches, and sliders of the mixing console 1221 todeliver the analog domain control settings to the mixing console 1221through the robotic arm 950. This user interface 1110 allows the user1271 to control the analog signal processors 1205 just as if he were inthe studio, but while being remotely located across a network 1203.Other user interfaces will be obvious to those of ordinary skill in theart having the benefit of this disclosure.

Turning now to FIG. 13, illustrated therein is one explanatory method1300 configured in accordance with one or more embodiments of thedisclosure. At step 1301, the method 1300 includes receiving, with aserver complex in communication with a network, a digital audio file andone or more analog domain control settings. In one or more embodiments,this step 1301 includes uploading, with the server complex, the digitalaudio file and the one or more analog domain control settings from aremote device across the network. For example, the remote device canupload the digital audio file to a cloud computer, from which the servercomplex can download the same.

At step 1302, the method 1300 includes converting, with a digital toanalog converter, the digital audio file to an analog signal. At step1303, the method 1300 includes adjusting one or more analog signalprocessors in accordance with the one or more analog domain controlsettings. At step optional step 1304, the method 1300 includesassigning, with the server complex, a clock frequency to the digitalaudio file that is different from a playback frequency used for normalplayback as previously described.

At step 1305, the method 1300 includes applying, with the one or moreanalog signal processors, at least one dynamic analog modification tothe analog signal. At step 1306, the method 1300 includes converting,with an analog to digital converter, the analog signal to a seconddigital audio file.

At optional step 1307, the method 1300 can include transmitting, withthe server complex across the network, one or more messages identifyinga mastering status of one or more of the digital audio file or thesecond digital audio file. For example, the method 800 may includesending one or more email messages, text messages, or othercommunications to the remote device to let a user know what step of themastering process their song is presently occurring.

At step 1308, the method 1300 includes delivering the second digitalaudio file to a remote device across the network. In one or moreembodiments, this includes storing the second digital audio file with acloud computer so that the remote device can download the same.

In the foregoing specification, specific embodiments of the presentdisclosure have been described. However, one of ordinary skill in theart appreciates that various modifications and changes can be madewithout departing from the scope of the present disclosure as set forthin the claims below. Thus, while preferred embodiments of the disclosurehave been illustrated and described, it is clear that the disclosure isnot so limited. Numerous modifications, changes, variations,substitutions, and equivalents will occur to those skilled in the artwithout departing from the spirit and scope of the present disclosure asdefined by the following claims. Accordingly, the specification andfigures are to be regarded in an illustrative rather than a restrictivesense, and all such modifications are intended to be included within thescope of present disclosure.

What is claimed is:
 1. An audio processing system, comprising: adigital-to-analog converter receiving a digital audio file andconverting the digital audio file to an analog signal; one or moreanalog signal processors receiving the analog signal from thedigital-to-analog converter; a robotic arm operable with the one or moreanalog signal processors, the robotic arm applying setting adjustmentsto the one or more analog signal processors in accordance with one ormore analog domain control settings; the one or more signal processorsapplying at least one analog modification to the analog signal inaccordance with the one or more analog domain control settings to theanalog signal to obtain a modified analog signal; and ananalog-to-digital converter converting the modified analog signal to amodified digital audio file.
 2. The audio processing system of claim 1,the robotic arm applying the setting adjustments to the one or moreanalog signal processors by adjusting one or more of knobs, sliders, orpotentiometers of the one or more analog signal processors in accordancewith the one or more analog domain control settings.
 3. The audioprocessing system of claim 1, the one or more analog signal processorscomprising a mixing console, the robotic arm adjusting the mixingconsoles in accordance with the one or more analog domain controlsettings.
 4. The audio processing system of claim 1, the robotic armincreasing compression settings of the one or more analog signalprocessors.
 5. The audio processing system of claim 1, the at least oneanalog modification comprising one or more of compression, limiting, orequalization.
 6. The audio processing system of claim 1, furthercomprising a first digital audio workstation comprising thedigital-to-analog converter.
 7. The audio processing system of claim 6,further comprising a second digital audio workstation comprising theanalog-to-digital converter.
 8. The audio processing system of claim 7,further comprising one or more control devices synchronizing delivery ofthe analog signal to the one or more analog signal processors and theconversion of the analog signal to the modified digital audio file. 9.The audio processing system of claim 8, the one or more control devicescausing a concurrent: initiation of a conversion of the digital audiofile to the analog signal at the first digital audio workstation todeliver the analog signal to the one or more analog signal processors;and conversion of the analog signal at the second digital audioworkstation to a second digital audio file after application of the atleast one analog modification.
 10. The audio processing system of claim1, the one or more analog domain control settings comprising a loudnesslevel associated with the digital audio file.
 11. The audio processingsystem of claim 1, further comprising a server complex receiving the oneor more analog domain control settings and delivering the one or moreanalog domain control settings to the digital-to-analog converter. 12.The audio processing system of claim 11, the server delivering a userinterface to a remote device across a network, the user interfacecomprising a virtual presentation of the one or more analog signalprocessors.
 13. The audio processing system of claim 12, the userinterface further comprising a network upload portal receiving an uploadof the digital audio file through the network upload portal.
 14. Theaudio processing system of claim 11, the server complex furthercomprising a network download portal facilitating download of themodified digital audio file across a network.
 15. A method, comprising:converting, with a digital to analog converter, a digital audio file toan analog signal; adjusting, with a robotic arm, one or more analogsignal processors receiving the analog signal from the digital to analogconverter in accordance with one or more analog domain control settings;applying, with the one or more analog signal processors, at least oneanalog modification to the analog signal; and converting, with an analogto digital converter, the analog signal to a second digital audio fileafter the applying.
 16. The method of claim 15, further comprisingreceiving, with a server complex, the digital audio file and the one ormore analog domain control settings from a remote device across anetwork.
 17. The method of claim 16, further comprising delivering thesecond digital audio file to the remote device across the network. 18.The method of claim 17, further comprising transmitting, with the servercomplex across the network, one or more messages identifying a masteringstatus of one or more of the digital audio file or the second digitalaudio file.
 19. The method of claim 17, further comprising assigning,with the server complex, a clock frequency to the digital audio filethat is different from a playback frequency used for normal playbackprior to the applying.
 20. The method of claim 15, further comprisingstoring one or more of the digital audio file or the second digitalaudio file with a cloud computer.